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‎_site/applications/index.html‎

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@@ -247,6 +247,8 @@ <h2 id="metering-and-analysis">Metering and Analysis</h2>
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<li><a href="http://www.baudline.com/"><strong>Baudline</strong></a>
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is a time-frequency browser designed for scientific visualization
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of the spectral domain.</li>
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<li><a href="http://humanleg.org.uk/code/brianalyze/"><strong>Brianalyze</strong></a>
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is a pretty simple JACK-connected spectrum analyzer</li>
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<li><a href="http://sourceforge.net/projects/brp-pacu"><strong>BRP-PACU</strong></a>
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is a dual channel FFT based acoustic analysis tool to help engineers
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configure professional sound systems by using the transfer function.</li>
@@ -385,9 +387,9 @@ <h2 id="programming-libraries">Programming Libraries</h2>
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<li><a href="http://clam-project.org/"><strong>Clam</strong></a>
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is a multiplatform software framework for
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research and application development for the Audio and Music domains.</li>
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<li><a href="http://faust.grame.fr/">**FAUST **</a>
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<li><a href="http://faust.grame.fr/"><strong>FAUST</strong></a>
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is a compiled language for real-time audio signal processing.</li>
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<li><a href="http://sourceforge.net/projects/gtkiostream/">GTKIOStream</a>
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<li><a href="http://sourceforge.net/projects/gtkiostream/"><strong>GTKIOStream</strong></a>
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provides allows easy use of JACK within GTK apps and has many other tools
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for DSP, Audio Masking, FFTs and plotting data.</li>
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<li><a href="http://x37v.info/projects/jackcpp"><strong>jackcpp</strong></a>
@@ -525,8 +527,8 @@ <h2 id="sound-generators">Sound Generators</h2>
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a software audio sampler.</li>
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<li><a href="http://www.selasky.org/hans_petter/midistudio"><strong>Midi Player Pro</strong></a>
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is the software you need to be able to play any kind of music
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in seconds with your fingertips. It uses [<strong>libumidi20</strong>]
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(http://www.freebsd.org/cgi/ports.cgi?query=libumidi&amp;stype=all&amp;sektion=all),
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in seconds with your fingertips. It uses
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<a href="http://www.freebsd.org/cgi/ports.cgi?query=libumidi&amp;stype=all&amp;sektion=all"><strong>libumidi20</strong></a>,
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which also uses Jack.</li>
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<li><a href="http://web.comhem.se/luna/"><strong>Mx44</strong></a>
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is a polyphonic multichannel midi realtime software synthesizer.</li>

‎_site/faq/comparing_jack.html‎

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@@ -44,86 +44,114 @@ <h2 id="project_tagline">TODO</h2>
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<h1 id="how-does-jack-compare-to-">How does JACK compare to …</h1>
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<h2 id="how-does-jack-compare-to">How does JACK compare to…?</h2>
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<p>Please mail the jack-devel mailing list if you have any concerns about
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the answers to these questions. Also, no disrespect to any effort is
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intended, only a recognition of different goals and design principles.</p>
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<p>Please mail the jackit-devel mailing list if you have any concerns about the
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answers to these questions. Also, no disrespect to any effort is intended,
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only a recognition of different goals and design principles.</p>
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<h3 id="other-linux-centered-systems">Other Linux-centered systems</h3>
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<h2 id="other-linux-centered-systems">Other Linux-centered systems</h2>
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<ul>
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<li>ALSA: both a HAL and a user-space library for audio under Linux.
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ALSA is used to provide the default audio i/o driver for JACK.
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ALSA is a very powerful audio API, but it does not provide a
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callback-based API or offer any solutions for inter-application
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communication, though it has been discussed and is theoretically possible.</li>
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<li>aRts, a streaming media architecture:
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<li>
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<p>ALSA: both a HAL and a user-space library for audio under Linux.
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ALSA is used to provide the default audio i/o driver for JACK. ALSA
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is a very powerful audio API, but it does not provide a callback-based
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API or offer any solutions for inter-application communication, though
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it has been discussed and is theoretically possible.</p>
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</li>
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<li>
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<p>aRts, a streaming media architecture:
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aRts was not designed from the ground up with low-latency in mind.
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Not a fault, but a design decision. A jack output element could be written
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for aRts, though, as far as I can tell.
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Note: aRts is not really used anymore by any Linux systems.</li>
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<li>GStreamer, another streaming media architecture:
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Not a fault, but a design decision. A jack output element could be
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written for aRts, though, as far as I can tell. Note: aRts is not
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really used anymore by any Linux systems.</p>
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</li>
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<li>
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<p>GStreamer, another streaming media architecture:
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GStreamer is designed for in-process construction of media pipelines,
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and is not used to link applications.
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JACK elements for GStreamer are under available.</li>
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<li>LADSPA, LV2: LADSPA is an internal plugin API for DSP routines,
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not a way of linking external applications together.</li>
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<li>Phonon</li>
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<li>Canberra</li>
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JACK elements for GStreamer are under available.</p>
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</li>
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<li>
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<p>LADSPA, LV2: LADSPA is an internal plugin API for DSP routines,
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not a way of linking external applications together.</p>
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</li>
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<li>
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<p>Phonon</p>
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</li>
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<li>
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<p>Canberra</p>
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</li>
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</ul>
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<h3 id="cross-platform-systems">Cross-platform systems</h3>
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<h2 id="cross-platform-systems">Cross-platform systems</h2>
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<ul>
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<li>PortAudio:
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a “cross platform, open-source, audio I/O library” offering both
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callback- and blocking I/O-based APIs.
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PortAudio backends exist for various Windows, Mac, and Unix HALs.
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It is mainly focused on hardware I/O rather than a general concept of
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ports and connections. The callback-style API used by both projects
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makes it relatively easy to port between the two (no pun intended),
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and there is a JACK backend for PortAudio so porting is not always necessary.</li>
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<li>SDL:</li>
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<li>SFML:</li>
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<li>OpenAL:</li>
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<li>
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<p>PortAudio:
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a “cross platform, open-source, audio I/O library” offering both
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callback- and blocking I/O-based APIs. PortAudio backends exist for
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various Windows, Mac, and Unix HALs. It is mainly focused on hardware
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I/O rather than a general concept of ports and connections. The
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callback-style API used by both projects makes it relatively easy to
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port between the two (no pun intended), and there is a JACK backend
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for PortAudio so porting is not always necessary.</p>
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</li>
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<li>
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<p>SDL:</p>
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</li>
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<li>
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<p>SFML:</p>
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</li>
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<li>
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<p>OpenAL:</p>
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</li>
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</ul>
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<h3 id="macos--andor-windows-centered-systems">MacOS- and/or Windows-centered systems</h3>
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<h2 id="macos--andor-windows-centered-systems">MacOS- and/or Windows-centered systems</h2>
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<ul>
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<li>CoreAudio, the Mac OS X audio API:
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Very similar to JACK in some the sense of being centered on
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a synchronous-execution-via-callback API, but does not include
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<li>
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<p>CoreAudio, the Mac OS X audio API:
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Very similar to JACK in some the sense of being centered on a
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synchronous-execution-via-callback API, but does not include
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inter-application audio routing. CoreAudio also includes a
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hardware-level abstraction layer, whereas JACK uses higher-level drivers
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for that purpose. The first JACK driver was based on ALSA,
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but others are available for the OSS and PortAudio interfaces.</li>
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<li>ASIO:
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a HAL for both Windows and MacOS that replaces the native device driver model
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with something much cleaner. It supports hardware-level latencies,
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but it does not connect applications to each other.
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Also, it is subject to license restrictions, and does not exist for Linux
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(though it would not be impossible to implement it on top of ALSA).</li>
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<li>ReWire, an inter-app communications API for Windows and MacOS
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from PropellerHeads and Steinberg, ReWire is similar in that it provides
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inter-application audio routing, but does not allow for
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fully independent processes, and has silly restrictions
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(“up to 64 channels”, etc).
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JACK also comes without silly license restrictions.</li>
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<li>VST, AudioUnits, DirectX, MAS, RTAS:
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hardware-level abstraction layer, whereas JACK uses higher-level
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drivers for that purpose. The first JACK driver was based on ALSA, but
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others are available for the OSS and PortAudio interfaces.</p>
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</li>
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<li>
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<p>ASIO:
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a HAL for both Windows and MacOS that replaces the native device
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driver model with something much cleaner. It supports hardware-level
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latencies, but it does not connect applications to each other. Also,
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it is subject to license restrictions, and does not exist for Linux
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(though it would not be impossible to implement it on top of ALSA).</p>
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</li>
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<li>
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<p>ReWire, an inter-app communications API for Windows and MacOS
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from PropellerHeads and Steinberg, ReWire is similar in that it
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provides inter-application audio routing, but does not allow for fully
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independent processes, and has silly restrictions (“up to 64
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channels”, etc). JACK also comes without silly license restrictions.</p>
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</li>
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<li>
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<p>VST, AudioUnits, DirectX, MAS, RTAS:
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these are all Windows/MacOS audio plugin APIs. None of them permit
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inter-application data sharing. Some plugin hosts can make this possible
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by using some other system such as ReWire. These APIs also require that
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the callback you write to process/generate data be executed
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in the context of the plugin host;
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JACK allows your callback to be executed within the context
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of your own application (if you wish to).</li>
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<li>Virtual Audio Cable, VAC (Windows only):
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inter-application data sharing. Some plugin hosts can make this
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possible by using some other system such as ReWire. These APIs also
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require that the callback you write to process/generate data be
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executed in the context of the plugin host; JACK allows your callback
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to be executed within the context of your own application (if you wish
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to).</p>
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</li>
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<li>
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<p>Virtual Audio Cable, VAC (Windows only):
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Creates a set of virtual audio devices named “Virtual Cables”, each
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consisting of a pair of the waveform input/output devices. Applications
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can send audio stream to an output side of a cable, and any other
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application can receive this stream from an input side. ASIO-compatible.</li>
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consisting of a pair of the waveform input/output devices.
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Applications can send audio stream to an output side of a cable, and
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any other application can receive this stream from an input side.
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ASIO-compatible.</p>
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</li>
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</ul>
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‎_site/faq/device_naming.html‎

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@@ -44,14 +44,15 @@ <h2 id="project_tagline">TODO</h2>
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<h1 id="how-do-i-refer-to-a-soundcard-so-that-the-name-always-works">How do I refer to a soundcard so that the name always works?</h1>
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<p>The precise order in which a Linux system discovers any soundcards can vary
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each time the machine is rebooted. This is particularly true of USB devices.
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The device that used to be “hw:0” can become “hw:1” after a reboot, or vice
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versa. Fortunately, you don’t need to use names like this, and can instead use
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a form that isn’t dependent on the order that the Linux discovers your
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soundcards.</p>
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<p>First, in a terminal window, run this command: <code class="highlighter-rouge">cat /proc/asound/cards</code>.
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<p>The precise order in which a Linux system discovers any soundcards can
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vary each time the machine is rebooted. This is particularly true of USB
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devices. The device that used to be <code class="highlighter-rouge">hw:0</code> can become <code class="highlighter-rouge">hw:1</code> after a
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reboot, or vice versa. Fortunately, you don’t need to use names like
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this, and can instead use a form that isn’t dependent on the order that
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the Linux discovers your soundcards.</p>
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<p>First, in a terminal window, run this command: <br />
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<code class="highlighter-rouge">cat /proc/asound/cards</code>. <br />
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You’ll see output a bit like this:</p>
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<div class="highlighter-rouge"><pre class="highlight"><code> 0 [SB ]: HDA-Intel - HDA ATI SB
@@ -66,19 +67,19 @@ <h1 id="how-do-i-refer-to-a-soundcard-so-that-the-name-always-works">How do I re
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<p>The “name” of each soundcard is in square brackets.</p>
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<p>With this information, you can now refer to a particular device as, for
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example <strong>hw:DSP</strong> (for the RME Hammerfall DSP in the above example. Even if
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the devices end up re-ordered for any reason, <strong>hw:DSP</strong> will still refer to
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the same device.</p>
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example <code class="highlighter-rouge">hw:DSP</code> (for the RME Hammerfall DSP in the above example.
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Even if the devices end up re-ordered for any reason, <code class="highlighter-rouge">hw:DSP</code> will
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still refer to the same device.</p>
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<p>The one case where this fails is if you have multiple instances of the same
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type of soundcard. This is a much trickier situation to deal with and is
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covered in detail <a href="http://alsa.opensrc.org/Udev">here</a>.</p>
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<p>The one case where this fails is if you have multiple instances of the
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same type of soundcard. This is a much trickier situation to deal with
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and is covered in detail <a href="http://alsa.opensrc.org/Udev">here</a>.</p>
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<h3 id="where-do-i-use-this-name">Where do I use this name?</h3>
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<p>You can type it into the device name selector in the upper left of qjacktl’s
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setup dialog. You can use it as the argument to the -d flag of JACK’s ALSA
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backend (e.g. <code class="highlighter-rouge">jackd -d alsa -d hw:DSP</code>.</p>
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<p>You can type it into the device name selector in the upper left of
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qjacktl’s setup dialog. You can use it as the argument to the <code class="highlighter-rouge">-d</code> flag
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of JACK’s ALSA backend (e.g. <code class="highlighter-rouge">jackd -d alsa -d hw:DSP</code>.</p>
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‎_site/faq/device_support.html‎

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@@ -50,14 +50,15 @@ <h2 id="on-os-x">On OS X</h2>
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<h2 id="on-linux">On Linux</h2>
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<p>All devices [supported by ALSA]
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(http://www.alsa-project.org/main/index.php/Matrix:Main)
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(PCI, USB and Bluetooth) or
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<a href="http://ffado.org/">FFADO</a> (FireWire) will work with JACK.</p>
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<p>All devices <a href="http://www.alsa-project.org/main/index.php/Matrix:Main">supported by
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ALSA</a> (PCI, USB
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and Bluetooth) or <a href="http://ffado.org/">FFADO</a> (FireWire) will work with
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JACK.</p>
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<h2 id="on-other-operating-systems">On other operating systems</h2>
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<p>If there is a working audio device driver then it will likely work with JACK.</p>
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<p>If there is a working audio device driver then it will likely work with
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JACK.</p>
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‎_site/faq/gstreamer_via_jack.html‎

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@@ -47,43 +47,45 @@ <h1 id="routing-gstreamer-audio-via-jack">Routing GStreamer audio via JACK</h1>
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<p>Its relatively easy to get audio from GStreamer to flow via JACK:</p>
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<ol>
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<li>You need the GStreamer JACK audio plugin, which is currently (Fall 2009)
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<li>
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<p>You need the GStreamer JACK audio plugin, which is currently (Fall 2009)
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part of the “bad” plugins collection. Most Linux distributions
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make this available through their normal software install/update systems.
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The name of the package containing this plugin will vary
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from distribution to distribution:
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on Fedora its called “gstreamer-plugins-bad-free-extra”,
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on Ubuntu its called “gst-plugins-bad”, etc. etc.</li>
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on Ubuntu its called “gst-plugins-bad”, etc. etc.</p>
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</li>
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<li>Next, you need to configure GStreamer to tell it to use this plugin
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for audio output. There are 3 methods available:
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<ul>
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<li>Through the command line using gconf2 (you may need to install this first)</li>
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<li>Through the command line using gconf2 (you may need to install this
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first)</li>
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<li>Install/run gconf-editor, which is a general purpose utility
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for configuring many GNOME-centric applications</li>
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for configuring many GNOME-centric applications.</li>
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<li>Some systems may have the gstreamer-properties command available,
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which offers a graphical tool for configuring gstreamer</li>
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which offers a graphical tool for configuring gstreamer.</li>
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</ul>
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</li>
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<li>Your goal is set the value of:
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<ul>
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<li>/system/gstreamer/0.10/audio/default/musicaudiosink</li>
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<li>/system/gstreamer/0.10/audio/default/audiosink</li>
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<li><code class="highlighter-rouge">/system/gstreamer/0.10/audio/default/musicaudiosink</code></li>
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<li><code class="highlighter-rouge">/system/gstreamer/0.10/audio/default/audiosink</code></li>
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</ul>
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</li>
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</ol>
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<p>to this value: <code class="highlighter-rouge">jackaudiosink buffer-time=2000000</code>. The exact value of
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<code class="highlighter-rouge">buffer-time</code> doesn’t matter too much, but higher values reduce the chance of
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glitches/drop-outs in the audio stream. You might also choose to set
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/system/gstreamer/0.10/audio/default/chataudiosink to the same value, but its
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less likely to be useful to you.</p>
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<p>The impact of this will vary depending on how much your system uses GStreamer.
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On modern GNOME-based systems, it will effectively divert all system sounds
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and all audio playback via JACK. On other systems (e.g. KDE-based systems) it
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will have less of an effect, though it will still impact specific applications
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that use GStreamer.</p>
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<code class="highlighter-rouge">buffer-time</code> doesn’t matter too much, but higher values reduce
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the chance of glitches/drop-outs in the audio stream. You might also
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choose to set <code class="highlighter-rouge">/system/gstreamer/0.10/audio/default/chataudiosink</code> to
82+
the same value, but its less likely to be useful to you.</p>
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<p>The impact of this will vary depending on how much your system uses
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GStreamer. On modern GNOME-based systems, it will effectively divert
86+
all system sounds and all audio playback via JACK. On other systems
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(e.g. KDE-based systems) it will have less of an effect, though it will
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still impact specific applications that use GStreamer.</p>
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</section>

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