Difference between revisions of "OpenAL Audio Server"

From Immersive Visualization Lab Wiki
Jump to: navigation, search
m (→‎Status)
(Updated to version 1.1)
 
(19 intermediate revisions by one user not shown)
Line 1: Line 1:
 
==Project Overview==
 
==Project Overview==
−
Rich, 3D environments need sound to provide the user a feeling of complete immersion. When visualization is provided by a cluster of computers, the sound processing can be handled by a dedicated machine that is in turn connected to the speakers and other audio equipment. The OpenAL Audio Server (OAS) provides this functionality via a network interface on a Linux machine. OAS is based loosely on the aging Windows DirectSound [http://www.calit2.net/~jschulze/projects/audioserver/ AudioServer] project by Marc Schreier, and aims to deliver a backwards compatible solution.
 
  
−
==Status==
+
Rich, 3D environments need sound to give the user a feeling of complete immersion. When
−
* '''Server Backend'''
+
visualization is provided by a cluster of computers, the sound processing can be handled by a  
−
-- The backend of the server is functional. Connections can be made, instructions and audio files can be sent, and audio can be played and stopped.<br>
+
single machine that is in turn connected to the speakers and other audio equipment. The
−
-- All network input and output is handled by a dedicated thread that parses the input into discrete audio instructions and queues it up. <br>
+
Open Audio Server (OAS) provides this functionality via a network interface on a Linux machine.
−
-- A separate thread retrieves these instructions from the queue and performs the desired action (i.e. play sound, move sound position, etc.) <br>
+
OAS is based loosely on the aging Windows DirectSound [http://www.calit2.net/~jschulze/projects/audioserver/ AudioServer]
 +
project by Marc Schreier, and is backwards compatible with all known client applications.
  
−
* '''Server Frontend'''
+
The OAS project is separated into two components: server and client. The server component contains
−
-- The frontend of the server uses FLTK and currently only consists of a log window. <br>
+
the source code and build system necessary to compile and run the actual audio server. The client
−
-- The frontend also resides in its own thread, allowing the user interface to operate independently from the backend of the server. <br>
+
component contains a client API to help applications communicate with the server. There is also
−
-- Errors are displayed in bold, warnings are displayed in italics, and informative messages contain no special formatting. <br>
+
sample code that can be compiled and played to demonstrate the server's functionality.
 +
 
 +
==Getting Started with the Server==
 +
 
 +
===Dependencies===
 +
 
 +
* [http://www.cmake.org/ CMake] (minimum v2.8) for the build system
 +
 
 +
* [http://kcat.strangesoft.net/openal.html OpenAL] for the backend audio processing
 +
 
 +
* [http://connect.creativelabs.com/openal/Downloads/ALUT/ ALUT] for file loading and OpenAL context management
 +
 
 +
* [http://www.minixml.org/ MXML (Minimal XML)] library for XML parsing
 +
 
 +
* Optional, but highly recommended: [http://www.fltk.org/ FLTK (v1.3.0)] for the GUI
 +
 
 +
* Optional: [http://www.stack.nl/~dimitri/doxygen/ Doxygen] (minimum 1.4.7) for documentation generation
 +
 
 +
===Building and Installing===
 +
 
 +
# Install the dependencies above, if you have not already done so
 +
# Get the OAS source code from the [https://github.com/CalVR/Open-Audio-Server GitHub repository:]<pre>git clone git://github.com/CalVR/Open-Audio-Server.git</pre>
 +
# Go to the <code>server/build</code> directory
 +
# If you want to configure build options yourself, do <pre>ccmake ..</pre> Otherwise, to go with the default options, you can do <pre>cmake ..</pre>
 +
# Assuming everything went okay with configuration, you can then compile with <pre>make</pre>
 +
#* With sudo priviledges, you can install with <pre>make install</pre>
 +
#* To generate documentation using Doxygen, use <pre>make docs</pre>
 +
 
 +
===Set-Up===
 +
 
 +
Before running the server, you will want to set some runtime options, such as which port you want
 +
the server to listen for connections on. Look at the sample configuration file `sample_oas_config.xml`
 +
to view this and other server options.
 +
 
 +
 
 +
===Running===
 +
 
 +
Pass the path to your XML configuration file as a command line argument to your OAS binary executable.
 +
<pre>OAS ~/oas_config.xml</pre>
 +
 
 +
 
 +
==Getting Started with the Client==
 +
 
 +
===Building and Installing===
 +
 
 +
# Get the OAS source code from the [https://github.com/CalVR/Open-Audio-Server GitHub repository:]<pre>git clone git://github.com/CalVR/Open-Audio-Server.git</pre>
 +
# Go to the <code>client/build</code> directory
 +
# If you want to configure build options yourself (such as compiling examples), do <pre>ccmake ..</pre> Otherwise, to go with the default options, you can do        <pre>cmake ..</pre>
 +
# Assuming everything went okay with configuration, you can then compile with <pre>make</pre>
 +
#* With sudo priviledges, you can install with <pre>make install</pre>
 +
#* To generate documentation using Doxygen, use <pre>make docs</pre>
 +
 
 +
===Examples===
 +
 
 +
There is a directory with source code containing basic examples that demonstrate the server's functionality.
 +
They are not compiled by default. The examples take various command line arguments, such as the IP address
 +
and port of the server the examples should connect to. By default, the examples will try to connect to
 +
the localhost, 127.0.0.1, at port 31231.
 +
 
 +
To enable compilation of the examples:
 +
# Go to the <code>client/build</code> directory
 +
# Configure build options using <pre>ccmake ..</pre>
 +
# Enable the <code>BUILD_EXAMPLES</code> option, and then reconfigure by pressing <code>'c'</code>
 +
# Enable the examples you want, and then reconfigure and generate the makefiles
 +
# Compile with <pre>make</pre>
 +
# Your examples should compile into the <code>client/build/examples</code> directory.
 +
 
 +
==Features==
 +
 
 +
All of the low level sound rendering is handled by OpenAL. There are two basic components to rendering
 +
sound: the listener, and the set of all individual sound sources.
 +
 
 +
===The Listener===
 +
The Listener represents the end-user inside the client application. For 3-D virtual environments, this
 +
is typically associated with the camera. The '''position''', '''orientation''', and '''velocity''' of the
 +
Listener will usually map to their corresponding counterparts for the camera.
 +
 
 +
===Sound Sources===
 +
Each sound source is associated with a particular sound file, and multiple sources can share the
 +
same sound file. These sound sources can also be moved around in the 3-D world. They all have
 +
their own '''position''', '''velocity''', and '''direction''', as well as other properties like gain
 +
or pitch. OAS does not set a hard limit to the number of sound sources that can be allocated at the
 +
same time. However, the underlying OpenAL implementation and sound card will most likely have a limit,
 +
(usually around ~256). As long as the client application releases unneeded sound sources in a timely
 +
manner, this limit should not be an issue. The server maps each sound source to a unique handle
 +
number, which is greater than or equal to 0. The client can then use this handle to reference the
 +
corresponding sound source.
 +
 
 +
====Multi-Channel vs Mono Sources====
 +
If a sound source is associated with a file that contains multiple channels of audio (such as stereo,
 +
or 5.1 surround), modifying the sound source's position, velocity, or direction will not have any
 +
effect. This is because multi-channel sources are assumed to be pre-mixed, and already contain
 +
all of the spatial information that is needed to render them in a 3D environment. Mono sources, or
 +
single-channel sources, can be thought of as a single point that emits sound, and therefore can be
 +
moved around in the 3D-world and rendered accurately. Although it is technically possible to handle
 +
positional audio rendering of multi-channel sound sources, it was not a primary goal of this project,
 +
and lies outside the scope of the desired feature set. Multi-channel/stereo sources can still be
 +
used very effectively for ambient/environmental audio. For example, stormy weather, forest noises,
 +
and background music are cases where the position of the sound is ambiguous or arbitrary. These can
 +
be great opportunities to use stereo or multi-channel premixed audio samples.
 +
 
 +
===The Doppler Effect===
 +
The doppler effect is when the pitch (frequency) of a sound source changes due to the sound source
 +
and the listener having a relative velocity to each other. In OpenAL and OAS, the '''velocity'''
 +
properties of both sound sources and the listener contribute only to the doppler shift calculations.
 +
As per the OpenAL specification, setting the '''velocity''' for the listener or sound sources will NOT
 +
cause the corresponding position property to be updated.
 +
 
 +
===The 3-D Coordinate System, and Units===
 +
The '''position''' properties of the listener and sound sources are specified in terms of 3-D world
 +
coordinates, <nowiki><</nowiki>'''x''', '''y''', '''z'''<nowiki>></nowiki>. '''Orientation''' of the
 +
listener and '''direction''' of sound sources are also vectors in the same world coordinate system.
 +
The units used for '''position''' can be arbitrary, as long as they are consistent between different
 +
sound sources and the listener. Although the '''velocity''' property is also a vector in the 3D world
 +
coordinate system, the units are completely independent from the units used for '''position'''.
 +
'''Velocity''', as mentioned [[#The_Doppler_Effect|above]], is only used for the Doppler Effect, and
 +
its units must be the same as the units used to define the speed of sound. By default, the
 +
'''speed of sound''' is defined as 343.3 meters per second. Therefore, by default, velocities need to
 +
be specified in terms of meters per second. The speed of sound is a global parameter that can be
 +
modified, as shown [[#Changing_Global_Parameters_of_Sound_Rendering|here]].
 +
 
 +
===Attenuation by Distance===
 +
OAS uses OpenAL's "inverse distance clamped" model to calculate the effective gain/volume of a sound based
 +
on the distance between the sound and the listener. This is equivalent to the IASIG I3DL2 model, and
 +
is also the default distance model in OpenAL. In short, the attenuation is calculated as follows:
 +
<br /> <br />
 +
<code>distance = Euclidean distance between sound source and listener</code> <br />
 +
<code>refDistance = the reference distance property of the sound source</code> <br />
 +
<code>rollOff = the rolloff factor property of the sound source</code> <br />
 +
<code>distance = MAX(distance, refDistance)</code> <br />
 +
<code>gain = refDistance / (refDistance + rollOff * (distance - refDistance))</code>
 +
<br /> <br />
 +
====Example of Distance Attenuation====
 +
Suppose you have two sounds - a helicopter, and someone talking (speech). The helicopter should be
 +
very audible in a wide surrounding area, whereas the speech should only be audible in a small area.
 +
To get the desired effect, and assuming units that are approximately in meters, we would want the
 +
reference distance of the helicopter to be ~25. On the other hand, the speech sound would have a
 +
reference distance of ~0.5. The rolloff factor for the helicopter would be in the range of
 +
~0.1, while the speech sound could have a rolloff factor between 1 and 2. The helicopter would then
 +
be audible even if the listener was far away, and the listener would have to be in the immediate
 +
vicinity of the speech source in order to hear it. Naturally, a wide range of other values can be
 +
used depending on the application, but the desired effect would be the same.
 +
 
 +
===Attenuation by Direction===
 +
When a sound source is directional, there are three important properties that determine how it
 +
is heard. Directional audio works by emulating a cone of sound. The effective gain of the sound
 +
changes based on where the listener is in this sound cone.
 +
 
 +
;Cone Inner Angle
 +
: The cone inner angle specifies another conical region inside the main sound cone, where the sound is not attenuated. If the listener falls within this region, it will hear the sound at the maximum possible gain, after accounting for distance. For the default non-directional sources, this value is essentially 360 degrees.
 +
;Cone Outer Angle
 +
: The cone outer angle specifies the outer, large conical region, where sound is attenuated based on the angular distance away from the central inner cone, as defined above. When the listener is close to the inner sound cone, the sound will have a higher effective gain. When the listener is far away from the inner sound cone and is near the extremities of the outer cone, the sound will be attenuated greatly, and will have a low effective gain.
 +
;Cone Outer Gain
 +
: The cone outer gain specifies the gain to be applied when the listener falls completely outside of the conical regions defined by the above angles. By default, this is 0, so that if the sound source is pointing completely away from a listener, the listener will not be able to hear the source.
 +
 
 +
==Client-Server Communication Protocol==
 +
 
 +
===Notes on Message Format===
 +
 
 +
The server uses a plain-text (ASCII) based communication protocol to receive instructions from the
 +
client. This allows the client to be even a simple telnet connection, if desired. The general format
 +
for each message passed from client to the server is a four letter, uppercase message identifier,
 +
followed by any relevant parameters. Integer and floating point numbers are also passed as ASCII
 +
plaintext.
 +
 
 +
Parameters to messages can be separated by commas, semicolons, or extra spaces. The server's parser
 +
is lenient with how parameters are separated. So,
 +
<pre>SSPO 4, 3.5, 6, 2.5</pre>
 +
is the same as
 +
<pre>SSPO 4 3.5 6 2.5</pre>
 +
is the same as
 +
<pre>SSPO 4; 3.5, 6, 2.5</pre>
 +
is the same as
 +
<pre>SSPO 4;;;;;  3.5,, ;6; 2.5</pre>
 +
However, it is recommended that you stay true to a convention. The provided client API uses the
 +
following convention, to minimize the number of bytes sent over the network:
 +
<pre>SSPO 4 3.5 6 2.5</pre>
 +
 
 +
===Multiple Messages per Packet===
 +
 
 +
The server can receive multiple messages per network packet. Each message should be separated by
 +
either a null character ('\0') or a whitespace character (' ', '\n', '\t').
 +
 
 +
'''Disclaimer''':
 +
Clients should not attempt to fill up network packets before transmitting them to the server,
 +
especially if the different messages all modify the same sound sources. There are two reasons:
 +
# The contents of each packet are processed very quickly, and many of the instructions in the packet may have no audible effect. For example, if the packet contains multiple instructions to change a sound source's position, only the very last of these instructions would have any kind of lasting effect that may be heard.
 +
# Attempting to fill up packets before transmission can introduce a potentially noticeable delay between the client application and the server. This is why the provided client API explicitly sets the TCP_NODELAY option on the socket.
 +
It is ultimately the client application's responsibility to find the balance between audio accuracy and network performance/congestion. The provided client API places a low priority on network congestion control, and will happily consume as much bandwidth as it needs to in order to keep up with the client application's demands.
 +
 
 +
It should be noted that even with TCP_NODELAY enabled, the network layer of the client API will automatically buffer multiple messages into one packet if the messages are being added faster than they can be transmitted.
 +
 
 +
===The Protocol===
 +
 
 +
====Creating and Releasing Sound Sources====
 +
 
 +
{| border="1" bgcolor="#FFFFFF"
 +
|-
 +
! scope="col" width="35%" | Message and Example(s)
 +
! scope="col" width="65%" | Description
 +
|-
 +
|
 +
GHDL filename
 +
<pre>GHDL beachsound.wav</pre>
 +
|
 +
Load '''filename''' into a sound source, and return a handle for accessing the source. If
 +
'''filename''' cannot be found in the server's cache directory, or a sound source cannot be
 +
created for '''filename''', the server will respond with "-1". Otherwise, the response will be a
 +
handle number, with values "0", "1", "2", "3", etc.
 +
|-
 +
|
 +
WAVE type frequency phase duration
 +
<pre>WAVE 1 261.3 0.0 3.5</pre>
 +
|
 +
Generate a new sound source based on the specified simple waveform. Similar to GHDL, the server
 +
will respond with a non-negative integer value in ASCII form on success, which will be the handle
 +
for the generated source. On failure, the server will respond with "-1".
 +
<br /> <br />
 +
The first parameter, '''type''', describes the shape of the wave that should be generated, and takes
 +
the following values:
 +
 
 +
{| border="1"
 +
|-
 +
! ''Type''
 +
! Wave Type
 +
|-
 +
! scope="row" align="center"| 1
 +
| Sinusoidal
 +
|-
 +
! scope="row" align="center"| 2
 +
| Square
 +
|-
 +
! scope="row" | 3
 +
| Sawtooth
 +
|-
 +
! scope="row" | 4
 +
| Whitenoise
 +
|-
 +
! scope="row" | 5
 +
| Impulse
 +
|}
 +
 
 +
'''Frequency''' specifies the [http://en.wikipedia.org/wiki/Audio_frequency frequency] of the waveform, and can be
 +
a floating point number.<br />
 +
'''Phase''' specifies the phase shift of the waveform, in degrees from -180 to +180, and can be a floating point.<br />
 +
'''Duration''' specifies how long the sound should last through one playback, in seconds, and can be a floating point.
 +
<br /> <br />
 +
 
 +
The example creates a sinusoidal wave, with a frequency corresponding to middle-C, a phase shift of
 +
0 degrees, and a duration of 3.5 seconds.
 +
|-
 +
|
 +
RHDL handle
 +
<pre>RHDL 1</pre>
 +
|
 +
Release the resources allocated for the source corresponding to '''handle'''.
 +
|}
 +
 
 +
====Controlling Playback====
 +
 
 +
{| border="1" bgcolor="#FFFFFF"
 +
|-
 +
! scope="col" width="35%" | Message and Example(s)
 +
! scope="col" width="65%" | Description
 +
|-
 +
|
 +
PLAY handle
 +
<pre>PLAY 5              </pre>
 +
|
 +
Play the source specified by '''handle'''. If the source is already playing, this will do nothing.
 +
|-
 +
|
 +
STOP handle
 +
<pre>STOP 5</pre>
 +
|
 +
Stop the source specified by '''handle'''. The playback position is reset to the beginning.
 +
|-
 +
|
 +
PAUS handle
 +
<pre>PAUS 5</pre>
 +
|
 +
Pause the source specified by '''handle'''. The current playback position is saved. To resume playback, PLAY must be used.
 +
Pausing a source that is already paused has no effect.
 +
|-
 +
|
 +
SSEC handle seconds
 +
<pre>SSEC 7 2.95</pre>
 +
|
 +
Set the '''playback position''', in seconds, of the source specified by '''handle'''. If the source is already playing,
 +
playback will skip to the desired location. Otherwise, the playback position will be applied next time the source is
 +
played. If the specified position is outside the bounds of the sound source, this will have no effect.
 +
<br /> <br />
 +
The example sets sound 7's play position to 2.95 seconds.
 +
|-
 +
|
 +
STAT handle
 +
<pre>STAT 4</pre>
 +
|
 +
Retrieve the current state of the sound specified by '''handle'''. The server responds with an integer ranging from
 +
'''0''' to '''5'''. Here are the possible values for the state:
 +
<br />
 +
 
 +
{| border="1"
 +
|-
 +
! ''Value''
 +
! ''State''
 +
! ''Description''
 +
|-
 +
! scope="row" align="center"| 0
 +
| Unknown/Error
 +
| The sound does not exist, or is invalid.
 +
|-
 +
! scope="row" align="center"| 1
 +
| Initial
 +
| The sound has a buffer associated with it, but the sound has never been played or used.
 +
|-
 +
! scope="row" | 2
 +
| Playing
 +
| The sound is currently playing.
 +
|-
 +
! scope="row" | 3
 +
| Paused
 +
| The sound is paused.
 +
|-
 +
! scope="row" | 4
 +
| Stopped
 +
| The sound is stopped.
 +
|-
 +
! scope="row" | 5
 +
| Deleted
 +
| The sound is being deleted, and will be invalid soon.
 +
|}
 +
|}
 +
 
 +
====Modifying Properties of Sound Sources====
 +
 
 +
{| border="1" bgcolor="#FFFFFF"
 +
|-
 +
! scope="col" width="35%" | Message and Example(s)
 +
! scope="col" width="65%" | Description
 +
 
 +
|-
 +
|
 +
SSPO handle x y z
 +
<pre>SSPO 3 4.5 0 22.337 </pre>
 +
|
 +
Set the position of the source specified by '''handle''' to <nowiki><</nowiki>'''x''', '''y''', '''z'''<nowiki>></nowiki>.
 +
The position values can be floating point, and have a default of <nowiki><</nowiki>0, 0, 0<nowiki>></nowiki> <br /> <br />
 +
 
 +
The example sets sound 3's position to <nowiki><4.5, 0, 22.337></nowiki>.
 +
|-
 +
|
 +
SSVE handle x y z
 +
<pre>SSVE 2 5.0 0 0</pre>
 +
|
 +
Set the velocity of the sound specified by '''handle''' to <nowiki><x, y, z></nowiki>. These values
 +
are only used for doppler effect calculations. OpenAL does not use the velocity for updating the
 +
sound's position, and OAS conforms to this specification. See [[#The_Doppler_Effect|the doppler effect ]]
 +
section for more information on how it works in OAS and OpenAL.
 +
<br /> <br />
 +
The example sets sound 2's velocity to 5.0 in the X direction.
 +
|-
 +
|
 +
SSVO handle gain
 +
<pre>SSVO 17 0.85</pre>
 +
|
 +
Set the gain (volume) of the sound specified by '''handle''' to '''gain'''. A gain of 0 will mute the sound
 +
completely, and the default gain for all sounds is 1. A gain of 0.5 corresponds to an attenuation of 6 dB. A gain
 +
value greater than 1.0 (to amplify the sound) is possible. However, the final gain value (after source-to-listener
 +
distance and orientation attentuation calculations) may be clamped by the sound card and drivers. If portability
 +
between systems is a key issue, it is not recommended to have widespread use of gain values greater than 1.  
 +
<br /> <br />
 +
The example sets sound 17's gain to 0.85, which will be quieter than the default.
 +
|-
 +
|
 +
FADE handle gain time
 +
<pre>FADE 12 0.7 4</pre>
 +
|
 +
Gradually and linearly change the sound's current gain value to the specified '''gain''' value, over the duration
 +
'''time''', in seconds. This can be used to slowly fade a sound's volume in or out, such as, but not limited to, when
 +
introducing objects/scenes or transitioning between different objects/scenes. Both '''gain''' and '''time''' can be
 +
floating point.
 +
<br /> <br />
 +
The example fades sound twelve's current gain value (whatever it may be) to 0.7, over the course of 4 seconds.
 +
|-
 +
|
 +
SSLP handle doLoop
 +
<pre>SSLP 3 1</pre>
 +
|
 +
Set the sound specified by '''handle''' to loop continuously, or to disable looping. The parameter '''doLoop''' takes
 +
boolean values of 0 or 1, with '''0''' disabling looping, and '''1''' enabling looping. A sound with looping enabled
 +
will resume playback from the beginning immediately after it is finished playing all the way through. Sounds have
 +
looping disabled by default.
 +
<br /> <br />
 +
The example turns looping on in sound 3.
 +
|-
 +
|
 +
SSDI handle x y z
 +
<pre>SSDI 3 0.5 0.0 -2.5</pre>
 +
|
 +
Set the direction of the sound specified by '''handle'''. The sound's direction is the vector specified by
 +
<nowiki><</nowiki>'''x''', '''y''', '''z'''<nowiki>></nowiki>. If the sound does not point towards the listener,
 +
the listener will not hear the sound at full volume. If the sound is pointing completely away from the listener, then
 +
it will not be audible at all. When the direction vector is the default of <nowiki><</nowiki>0,0,0<nowiki>></nowiki>,
 +
the sound source has no direction associated with it. Sound is then emitted equally in all directions, similarly to
 +
a point light source.
 +
<br /> <br />
 +
The example sets the direction of sound 3 to <nowiki><</nowiki>0.5, 0.0, -2.5<nowiki>></nowiki>.
 +
|-
 +
|
 +
SSDI handle angle
 +
<pre>SSDI 3 2.944</pre>
 +
|
 +
Set the direction of the sound specified by '''handle''' using '''angle''', in radians. The angle is converted to a unit
 +
vector in the X-Z plane using sine and cosine. There is no default angle value, because sound sources are not
 +
directional by default.. Once you use this version of SSDI on a sound source, the sound source will remain directional
 +
until you use "SSDI handle 0 0 0", to fully disable the directionality.
 +
<br /> <br />
 +
The example has the same result as the example for SSDI with a vector parameter. The direction of sound 3 is set to
 +
2.944 radians.
 +
|-
 +
|
 +
SSDV handle angle gain
 +
<pre>SSDV 3 2.944 0.869</pre>
 +
|
 +
Set both the direction and gain of the sound given by '''handle'''. The direction is specified by '''angle''', in radians,
 +
and the volume is set by '''gain'''. This effectively combines two messages: SSDI with the angle parameter and SSVO.
 +
<br /> <br />
 +
The example sets the direction and gain of sound 3 to 2.944 and 0.869, respectively.
 +
|-
 +
|
 +
SPIT handle pitch
 +
<pre>SPIT 3 1.25</pre>
 +
|
 +
Set the pitch of sound specified by '''handle'''. '''Pitch''' can be any floating point greater than 0. A value of 1 is
 +
the default. Doubling the pitch will increase the pitch of the sound by one octave, and halving the pitch will drop the
 +
sound by one octave. Modifying the pitch also changes the playback speed, so a sound with pitch greater than 1 will play
 +
at a faster speed, and a sound with a pitch less than 1 will play at a slower speed.
 +
<br /> <br />
 +
The example sets the pitch of sound 3 to 1.25.
 +
|-
 +
|
 +
SPAR handle parameter value
 +
<pre>SPAR 8 1 0.001      </pre>
 +
<pre>SPAR 8 2 100        </pre>
 +
|
 +
This instruction can be used to set sound rendering parameters for the sound source specified by '''handle'''. Currently,
 +
the following parameters are available:
 +
<br />
 +
 
 +
{| border="1"
 +
|-
 +
! ''Parameter''
 +
! ''Value''
 +
! ''Description''
 +
|-
 +
! scope="row" align="center"| Rolloff Factor
 +
| style="text-align: center;" | 1
 +
|
 +
The rolloff factor is used to control the rate at which sounds attenuate with respect to their distance away from
 +
the listener.
 +
<br /> <br />
 +
A higher rolloff factor will result in sounds becoming quieter faster, as the distance between the sound and the listener
 +
is increased. A rolloff of 0 will mean that sounds do not attenuate with respect to distance (the sounds will seem just
 +
as loud no matter how far away they are from the listener). For a client application using distance units of millimeters
 +
(so that position coordinates are often in magnitude of thousands), a rolloff factor of 0.001 is likely going to be
 +
appropriate. See the [[#Attenuation_by_Distance|attenuation section]]
 +
for more information.
 +
|-
 +
! scope="row" align="center"| Reference Distance
 +
| style="text-align: center;" | 2
 +
|
 +
The reference distance describes the radius for a sphere around a sound source in which the gain is not attenuated based on
 +
distance. If the distance between the sound and the listener is less than or equal to the reference distance, the sound's
 +
gain will not be attenuated (reduced) based on how far the sound is from the listener. Values must be greater than or equal
 +
to 0. See the [[#Attenuation_by_Distance|attenuation section]]
 +
for more information.
 +
|}
 +
<br /> <br />
 +
 
 +
The first example sets the rolloff factor of sound source 8 to 0.001, which is appropriate for generic sounds and positional units
 +
of millimeters.
 +
<br /> <br />
 +
The second example sets the reference distance of sound source 8 to 100.
 +
|}
 +
 
 +
 
 +
====Modifying Properties of the Listener====
 +
 
 +
{| border="1" bgcolor="#FFFFFF"
 +
|-
 +
! scope="col" width="35%" | Message and Example(s)
 +
! scope="col" width="65%" | Description
 +
|-
 +
|
 +
GAIN value
 +
<pre>GAIN 0.0</pre>
 +
|
 +
Change the gain (volume) for the listener to '''value'''. The default gain for the listener is 1.0. This can be useful
 +
especially to mute all sounds for the listener, as the example demonstrates.
 +
|-
 +
|
 +
SLPO x y z
 +
<pre>SLPO 20.1 0.08 -9.23</pre>
 +
|
 +
Set the listener's position to <nowiki><</nowiki>'''x''', '''y''', '''z'''<nowiki>></nowiki>, in the same coordinate
 +
system as the sound sources. The default position is <nowiki><</nowiki>0, 0, 0<nowiki>></nowiki>
 +
<br /> <br />
 +
The example sets the position to <nowiki><</nowiki>20.1, 0.08, -9.23<nowiki>></nowiki>.
 +
|-
 +
|
 +
SLOR aX aY aZ uX uY uZ
 +
<pre>SLOR 1 0 0 0 1 0</pre>
 +
|
 +
Set the listener's orientation using two vectors. The first,
 +
<nowiki><</nowiki>'''aX''', '''aY''', '''aZ'''<nowiki>></nowiki>, specifies the "look-at" direction. The second,
 +
<nowiki><</nowiki>'''uX''', '''uY''', '''uZ'''<nowiki>></nowiki>, specifies the "up" direction. If two vectors are linearly
 +
dependent, the behavior is undefined. They do not need to be normalized. The defaults are
 +
<nowiki><</nowiki>0, 0, -1<nowiki>></nowiki> for the look-at vector and <nowiki><</nowiki>0, 1, 0<nowiki>></nowiki> for the
 +
up vector.
 +
<br /> <br />
 +
The example sets the listener's orientation to <nowiki><</nowiki>1, 0, 0<nowiki>></nowiki> for the look-at vector and
 +
<nowiki><</nowiki>0, 1, 0<nowiki>></nowiki> for the up vector.
 +
|-
 +
|
 +
SLVE x y z
 +
<pre>SLVE -2.0 0 0</pre>
 +
|
 +
Set the listener's velocity to <nowiki><</nowiki>'''x''', '''y''', '''z'''<nowiki>></nowiki>. Similar to setting a sound source's
 +
velocity, the server does not update the position of the listener based on this specified velocity. It is only used as a
 +
parameter for doppler effect calculations. See [[#The_Doppler_Effect|the doppler effect]]
 +
section for more information.
 +
<br /> <br />
 +
The example sets the velocity of the listener to -2.0 in the X direction.
 +
|}
 +
 
 +
====Changing Global Parameters of Sound Rendering====
 +
{| border="1" bgcolor="#FFFFFF"
 +
|-
 +
! scope="col" width="35%" | Message and Example(s)
 +
! scope="col" width="65%" | Description
 +
|-
 +
|
 +
PARA parameter value
 +
<pre>PARA 1 343300      </pre>
 +
<pre>PARA 3 0.001        </pre>
 +
|
 +
This instruction can be used to set various global sound rendering parameters. Currently, the following parameters are available:
 +
<br />
 +
 
 +
{| border="1"
 +
|-
 +
! ''Parameter''
 +
! ''Value''
 +
! ''Description''
 +
|-
 +
! scope="row" align="center"| Speed of Sound
 +
| style="text-align: center;" | 1
 +
|
 +
The speed of sound is used for doppler effect calculations, and must be in the same units as the velocities of listener and
 +
sound sources. The default value is 343.3 meters per second, appropriate for the speed of sound through air.
 +
|-
 +
! scope="row" align="center"| Doppler Factor
 +
| style="text-align: center;" | 2
 +
|
 +
The doppler factor can be used to emphasize or de-emphasize the pitch shifts due to the doppler effect. The default value
 +
is 1.0, and a value of 0 will effectively disable any pitch shifts due to the doppler effect.
 +
|-
 +
! scope="row" align="center"| Default Rolloff Factor
 +
| style="text-align: center;" | 3
 +
|
 +
The default rolloff factor is used to control the rate at which sounds attenuate with respect to their distance away from
 +
the listener. NOTE: This property is not applied to existing sources - it only affects sources generated in the future.
 +
This parameter is provided so that the client can conveniently set the scale of the world during the initialization of
 +
their program.
 +
<br /> <br />
 +
A higher rolloff factor will result in sounds becoming quieter faster, as the distance between the sound and the listener
 +
is increased. A rolloff of 0 will mean that sounds do not attenuate with respect to distance (the sounds will seem just
 +
as loud no matter how far away they are from the listener). For a client application using distance units of millimeters
 +
(so that position coordinates are often in magnitude of thousands), a rolloff factor of 0.001 is likely going to be
 +
appropriate. See the [[#Attenuation_by_Distance|attenuation section]]
 +
for more information.
 +
|-
 +
! scope="row" align="center"| Default Reference Distance
 +
| style="text-align: center;" | 4
 +
|
 +
The reference distance describes a sphere around a sound source in which the gain is not attenuated based on distance.
 +
If the distance between the sound and the listener is less than or equal to the reference distance, the sound's
 +
gain will not be attenuated (reduced) based on how far the sound is from the listener. Values must be greater than or
 +
equal to 0. See the [[#Attenuation_by_Distance|attenuation section]]
 +
for more information.
 +
<br /> <br />
 +
NOTE: This property is not applied to existing sources - it only affects sources generated in the future.
 +
This parameter is provided so that the client can conveniently configure the scale of the world during the initialization
 +
of their program.
 +
|}
 +
<br /> <br />
 +
 
 +
The first example sets the speed of sound to 343300, corresponding to the units of millimeters per second,
 +
through the medium of air.
 +
<br /> <br />
 +
The second example sets the default rolloff factor to 0.001, which is appropriate for generic sounds and positional units
 +
of millimeters.
 +
|}
 +
 
 +
====Deprecated Messages====
 +
{| border="1" bgcolor="#FFFFFF"
 +
|-
 +
! scope="col" width="35%" | Message and Example(s)
 +
! scope="col" width="65%" | Description
 +
|-
 +
|
 +
SSVE handle speed
 +
<pre>SSVE 6 3.5          </pre>
 +
|
 +
('''''deprecated''''') <br />
 +
Set the sound's speed. The sound's current direction is used to compute the effective velocity. This assumes
 +
that the sound is moving in the same direction it is facing. Note that if the sound has no direction associated with it, the
 +
behavior is undefined. This message is leftover to support old applications that were tailored to the old Windows AudioServer
 +
project. New applications should avoid using this functionality, or use at their own risk.
 +
<br /> <br />
 +
The example sets sound 6's speed to 3.5.
 +
 
 +
|}
 +
 
 +
====Unsupported Messages====
 +
 
 +
There were some messages that had inconsistent, incomplete, or altogether broken support in the old Windows AudioServer.
 +
This meant that in order to maintain full backwards compatibility, OAS would have had to implement these deprecated
 +
messages in a similar, broken manner. Instead, support for these messages has been dropped completely, under the
 +
assumption that old client applications would not be dependent on something that was broken. Support may be added in
 +
the future as needed.
 +
 
 +
{| border="1" bgcolor="#FFFFFF"
 +
|- align="center"
 +
|
 +
'''Messages that are no longer supported:'''
 +
|-
 +
|
 +
<pre>SSDR handle angle</pre>
 +
|-
 +
|
 +
<pre>SSRV handle angle gain</pre>
 +
|-
 +
|
 +
<pre>SSRV handle x y z gain</pre>
 +
|}
  
 
==To-Do==
 
==To-Do==
−
* Bullet Point
+
* Extend the [http://code.google.com/p/osgaudio/ osgAudio nodekit] to add support for the OAS client API
  
 
==Future Work==
 
==Future Work==
−
* Bullet Point
+
* Enhance sockets code to add support for multiple connected clients. Asynchronous socket I/O.
  
 
==Participants==
 
==Participants==
Line 29: Line 675:
 
* [http://www.calit2.net/~jschulze/ Jurgen Schulze]
 
* [http://www.calit2.net/~jschulze/ Jurgen Schulze]
  
−
Development Assistance:
+
Misc. Development Assistance:
  
 
* Philip Weber
 
* Philip Weber
 +
* Andrew Prudhomme
  
  
−
Initial Concept Base:
+
Initial Base Concept:
  
 
* [http://www.calit2.net/~jschulze/projects/audioserver/ AudioServer Project by Marc Schreier]
 
* [http://www.calit2.net/~jschulze/projects/audioserver/ AudioServer Project by Marc Schreier]

Latest revision as of 13:51, 13 August 2012

Contents

Project Overview

Rich, 3D environments need sound to give the user a feeling of complete immersion. When visualization is provided by a cluster of computers, the sound processing can be handled by a single machine that is in turn connected to the speakers and other audio equipment. The Open Audio Server (OAS) provides this functionality via a network interface on a Linux machine. OAS is based loosely on the aging Windows DirectSound AudioServer project by Marc Schreier, and is backwards compatible with all known client applications.

The OAS project is separated into two components: server and client. The server component contains the source code and build system necessary to compile and run the actual audio server. The client component contains a client API to help applications communicate with the server. There is also sample code that can be compiled and played to demonstrate the server's functionality.

Getting Started with the Server

Dependencies

  • CMake (minimum v2.8) for the build system
  • OpenAL for the backend audio processing
  • ALUT for file loading and OpenAL context management
  • Optional: Doxygen (minimum 1.4.7) for documentation generation

Building and Installing

  1. Install the dependencies above, if you have not already done so
  2. Get the OAS source code from the GitHub repository:
    git clone git://github.com/CalVR/Open-Audio-Server.git
  3. Go to the server/build directory
  4. If you want to configure build options yourself, do 
    ccmake ..
    Otherwise, to go with the default options, you can do 
    cmake ..
  5. Assuming everything went okay with configuration, you can then compile with 
    make
    • With sudo priviledges, you can install with 
      make install
    • To generate documentation using Doxygen, use 
      make docs

Set-Up

Before running the server, you will want to set some runtime options, such as which port you want the server to listen for connections on. Look at the sample configuration file `sample_oas_config.xml` to view this and other server options.


Running

Pass the path to your XML configuration file as a command line argument to your OAS binary executable. 

OAS ~/oas_config.xml


Getting Started with the Client

Building and Installing

  1. Get the OAS source code from the GitHub repository:
    git clone git://github.com/CalVR/Open-Audio-Server.git
  2. Go to the client/build directory
  3. If you want to configure build options yourself (such as compiling examples), do 
    ccmake ..
    Otherwise, to go with the default options, you can do 
    cmake ..
  4. Assuming everything went okay with configuration, you can then compile with 
    make
    • With sudo priviledges, you can install with 
      make install
    • To generate documentation using Doxygen, use 
      make docs

Examples

There is a directory with source code containing basic examples that demonstrate the server's functionality. They are not compiled by default. The examples take various command line arguments, such as the IP address and port of the server the examples should connect to. By default, the examples will try to connect to the localhost, 127.0.0.1, at port 31231.

To enable compilation of the examples:

  1. Go to the client/build directory
  2. Configure build options using 
    ccmake ..
  3. Enable the BUILD_EXAMPLES option, and then reconfigure by pressing 'c'
  4. Enable the examples you want, and then reconfigure and generate the makefiles
  5. Compile with 
    make
  6. Your examples should compile into the client/build/examples directory.

Features

All of the low level sound rendering is handled by OpenAL. There are two basic components to rendering sound: the listener, and the set of all individual sound sources.

The Listener

The Listener represents the end-user inside the client application. For 3-D virtual environments, this is typically associated with the camera. The position, orientation, and velocity of the Listener will usually map to their corresponding counterparts for the camera.

Sound Sources

Each sound source is associated with a particular sound file, and multiple sources can share the same sound file. These sound sources can also be moved around in the 3-D world. They all have their own position, velocity, and direction, as well as other properties like gain or pitch. OAS does not set a hard limit to the number of sound sources that can be allocated at the same time. However, the underlying OpenAL implementation and sound card will most likely have a limit, (usually around ~256). As long as the client application releases unneeded sound sources in a timely manner, this limit should not be an issue. The server maps each sound source to a unique handle number, which is greater than or equal to 0. The client can then use this handle to reference the corresponding sound source.

Multi-Channel vs Mono Sources

If a sound source is associated with a file that contains multiple channels of audio (such as stereo, or 5.1 surround), modifying the sound source's position, velocity, or direction will not have any effect. This is because multi-channel sources are assumed to be pre-mixed, and already contain all of the spatial information that is needed to render them in a 3D environment. Mono sources, or single-channel sources, can be thought of as a single point that emits sound, and therefore can be moved around in the 3D-world and rendered accurately. Although it is technically possible to handle positional audio rendering of multi-channel sound sources, it was not a primary goal of this project, and lies outside the scope of the desired feature set. Multi-channel/stereo sources can still be used very effectively for ambient/environmental audio. For example, stormy weather, forest noises, and background music are cases where the position of the sound is ambiguous or arbitrary. These can be great opportunities to use stereo or multi-channel premixed audio samples.

The Doppler Effect

The doppler effect is when the pitch (frequency) of a sound source changes due to the sound source and the listener having a relative velocity to each other. In OpenAL and OAS, the velocity properties of both sound sources and the listener contribute only to the doppler shift calculations. As per the OpenAL specification, setting the velocity for the listener or sound sources will NOT cause the corresponding position property to be updated.

The 3-D Coordinate System, and Units

The position properties of the listener and sound sources are specified in terms of 3-D world coordinates, <x, y, z>. Orientation of the listener and direction of sound sources are also vectors in the same world coordinate system. The units used for position can be arbitrary, as long as they are consistent between different sound sources and the listener. Although the velocity property is also a vector in the 3D world coordinate system, the units are completely independent from the units used for position. Velocity, as mentioned above, is only used for the Doppler Effect, and its units must be the same as the units used to define the speed of sound. By default, the speed of sound is defined as 343.3 meters per second. Therefore, by default, velocities need to be specified in terms of meters per second. The speed of sound is a global parameter that can be modified, as shown here.

Attenuation by Distance

OAS uses OpenAL's "inverse distance clamped" model to calculate the effective gain/volume of a sound based on the distance between the sound and the listener. This is equivalent to the IASIG I3DL2 model, and is also the default distance model in OpenAL. In short, the attenuation is calculated as follows:

distance = Euclidean distance between sound source and listener
refDistance = the reference distance property of the sound source
rollOff = the rolloff factor property of the sound source
distance = MAX(distance, refDistance)
gain = refDistance / (refDistance + rollOff * (distance - refDistance))

Example of Distance Attenuation

Suppose you have two sounds - a helicopter, and someone talking (speech). The helicopter should be very audible in a wide surrounding area, whereas the speech should only be audible in a small area. To get the desired effect, and assuming units that are approximately in meters, we would want the reference distance of the helicopter to be ~25. On the other hand, the speech sound would have a reference distance of ~0.5. The rolloff factor for the helicopter would be in the range of ~0.1, while the speech sound could have a rolloff factor between 1 and 2. The helicopter would then be audible even if the listener was far away, and the listener would have to be in the immediate vicinity of the speech source in order to hear it. Naturally, a wide range of other values can be used depending on the application, but the desired effect would be the same.

Attenuation by Direction

When a sound source is directional, there are three important properties that determine how it is heard. Directional audio works by emulating a cone of sound. The effective gain of the sound changes based on where the listener is in this sound cone.

Cone Inner Angle
The cone inner angle specifies another conical region inside the main sound cone, where the sound is not attenuated. If the listener falls within this region, it will hear the sound at the maximum possible gain, after accounting for distance. For the default non-directional sources, this value is essentially 360 degrees.
Cone Outer Angle
The cone outer angle specifies the outer, large conical region, where sound is attenuated based on the angular distance away from the central inner cone, as defined above. When the listener is close to the inner sound cone, the sound will have a higher effective gain. When the listener is far away from the inner sound cone and is near the extremities of the outer cone, the sound will be attenuated greatly, and will have a low effective gain.
Cone Outer Gain
The cone outer gain specifies the gain to be applied when the listener falls completely outside of the conical regions defined by the above angles. By default, this is 0, so that if the sound source is pointing completely away from a listener, the listener will not be able to hear the source.

Client-Server Communication Protocol

Notes on Message Format

The server uses a plain-text (ASCII) based communication protocol to receive instructions from the client. This allows the client to be even a simple telnet connection, if desired. The general format for each message passed from client to the server is a four letter, uppercase message identifier, followed by any relevant parameters. Integer and floating point numbers are also passed as ASCII plaintext.

Parameters to messages can be separated by commas, semicolons, or extra spaces. The server's parser is lenient with how parameters are separated. So, 

SSPO 4, 3.5, 6, 2.5

is the same as 

SSPO 4 3.5 6 2.5

is the same as 

SSPO 4; 3.5, 6, 2.5

is the same as 

SSPO 4;;;;;  3.5,, ;6; 2.5

However, it is recommended that you stay true to a convention. The provided client API uses the following convention, to minimize the number of bytes sent over the network: 

SSPO 4 3.5 6 2.5

Multiple Messages per Packet

The server can receive multiple messages per network packet. Each message should be separated by either a null character ('\0') or a whitespace character (' ', '\n', '\t').

Disclaimer: Clients should not attempt to fill up network packets before transmitting them to the server, especially if the different messages all modify the same sound sources. There are two reasons:

  1. The contents of each packet are processed very quickly, and many of the instructions in the packet may have no audible effect. For example, if the packet contains multiple instructions to change a sound source's position, only the very last of these instructions would have any kind of lasting effect that may be heard.
  2. Attempting to fill up packets before transmission can introduce a potentially noticeable delay between the client application and the server. This is why the provided client API explicitly sets the TCP_NODELAY option on the socket.

It is ultimately the client application's responsibility to find the balance between audio accuracy and network performance/congestion. The provided client API places a low priority on network congestion control, and will happily consume as much bandwidth as it needs to in order to keep up with the client application's demands.

It should be noted that even with TCP_NODELAY enabled, the network layer of the client API will automatically buffer multiple messages into one packet if the messages are being added faster than they can be transmitted.

The Protocol

Creating and Releasing Sound Sources

Message and Example(s) Description

GHDL filename 

GHDL beachsound.wav

Load filename into a sound source, and return a handle for accessing the source. If filename cannot be found in the server's cache directory, or a sound source cannot be created for filename, the server will respond with "-1". Otherwise, the response will be a handle number, with values "0", "1", "2", "3", etc.

WAVE type frequency phase duration 

WAVE 1 261.3 0.0 3.5

Generate a new sound source based on the specified simple waveform. Similar to GHDL, the server will respond with a non-negative integer value in ASCII form on success, which will be the handle for the generated source. On failure, the server will respond with "-1".

The first parameter, type, describes the shape of the wave that should be generated, and takes the following values:

Type Wave Type
1 Sinusoidal
2 Square
3 Sawtooth
4 Whitenoise
5 Impulse

Frequency specifies the frequency of the waveform, and can be a floating point number.
Phase specifies the phase shift of the waveform, in degrees from -180 to +180, and can be a floating point.
Duration specifies how long the sound should last through one playback, in seconds, and can be a floating point.

The example creates a sinusoidal wave, with a frequency corresponding to middle-C, a phase shift of 0 degrees, and a duration of 3.5 seconds.

RHDL handle 

RHDL 1

Release the resources allocated for the source corresponding to handle.

Controlling Playback

Message and Example(s) Description

PLAY handle 

PLAY 5

Play the source specified by handle. If the source is already playing, this will do nothing.

STOP handle 

STOP 5

Stop the source specified by handle. The playback position is reset to the beginning.

PAUS handle 

PAUS 5

Pause the source specified by handle. The current playback position is saved. To resume playback, PLAY must be used. Pausing a source that is already paused has no effect.

SSEC handle seconds 

SSEC 7 2.95

Set the playback position, in seconds, of the source specified by handle. If the source is already playing, playback will skip to the desired location. Otherwise, the playback position will be applied next time the source is played. If the specified position is outside the bounds of the sound source, this will have no effect.

The example sets sound 7's play position to 2.95 seconds.

STAT handle 

STAT 4

Retrieve the current state of the sound specified by handle. The server responds with an integer ranging from 0 to 5. Here are the possible values for the state:

Value State Description
0 Unknown/Error The sound does not exist, or is invalid.
1 Initial The sound has a buffer associated with it, but the sound has never been played or used.
2 Playing The sound is currently playing.
3 Paused The sound is paused.
4 Stopped The sound is stopped.
5 Deleted The sound is being deleted, and will be invalid soon.

Modifying Properties of Sound Sources

Message and Example(s) Description

SSPO handle x y z 

SSPO 3 4.5 0 22.337

Set the position of the source specified by handle to <x, y, z>. The position values can be floating point, and have a default of <0, 0, 0>

The example sets sound 3's position to <4.5, 0, 22.337>.

SSVE handle x y z 

SSVE 2 5.0 0 0

Set the velocity of the sound specified by handle to <x, y, z>. These values are only used for doppler effect calculations. OpenAL does not use the velocity for updating the sound's position, and OAS conforms to this specification. See the doppler effect section for more information on how it works in OAS and OpenAL.

The example sets sound 2's velocity to 5.0 in the X direction.

SSVO handle gain 

SSVO 17 0.85

Set the gain (volume) of the sound specified by handle to gain. A gain of 0 will mute the sound completely, and the default gain for all sounds is 1. A gain of 0.5 corresponds to an attenuation of 6 dB. A gain value greater than 1.0 (to amplify the sound) is possible. However, the final gain value (after source-to-listener distance and orientation attentuation calculations) may be clamped by the sound card and drivers. If portability between systems is a key issue, it is not recommended to have widespread use of gain values greater than 1.

The example sets sound 17's gain to 0.85, which will be quieter than the default.

FADE handle gain time 

FADE 12 0.7 4

Gradually and linearly change the sound's current gain value to the specified gain value, over the duration time, in seconds. This can be used to slowly fade a sound's volume in or out, such as, but not limited to, when introducing objects/scenes or transitioning between different objects/scenes. Both gain and time can be floating point.

The example fades sound twelve's current gain value (whatever it may be) to 0.7, over the course of 4 seconds.

SSLP handle doLoop 

SSLP 3 1

Set the sound specified by handle to loop continuously, or to disable looping. The parameter doLoop takes boolean values of 0 or 1, with 0 disabling looping, and 1 enabling looping. A sound with looping enabled will resume playback from the beginning immediately after it is finished playing all the way through. Sounds have looping disabled by default.

The example turns looping on in sound 3.

SSDI handle x y z 

SSDI 3 0.5 0.0 -2.5

Set the direction of the sound specified by handle. The sound's direction is the vector specified by <x, y, z>. If the sound does not point towards the listener, the listener will not hear the sound at full volume. If the sound is pointing completely away from the listener, then it will not be audible at all. When the direction vector is the default of <0,0,0>, the sound source has no direction associated with it. Sound is then emitted equally in all directions, similarly to a point light source.

The example sets the direction of sound 3 to <0.5, 0.0, -2.5>.

SSDI handle angle 

SSDI 3 2.944

Set the direction of the sound specified by handle using angle, in radians. The angle is converted to a unit vector in the X-Z plane using sine and cosine. There is no default angle value, because sound sources are not directional by default.. Once you use this version of SSDI on a sound source, the sound source will remain directional until you use "SSDI handle 0 0 0", to fully disable the directionality.

The example has the same result as the example for SSDI with a vector parameter. The direction of sound 3 is set to 2.944 radians.

SSDV handle angle gain 

SSDV 3 2.944 0.869

Set both the direction and gain of the sound given by handle. The direction is specified by angle, in radians, and the volume is set by gain. This effectively combines two messages: SSDI with the angle parameter and SSVO.

The example sets the direction and gain of sound 3 to 2.944 and 0.869, respectively.

SPIT handle pitch 

SPIT 3 1.25

Set the pitch of sound specified by handle. Pitch can be any floating point greater than 0. A value of 1 is the default. Doubling the pitch will increase the pitch of the sound by one octave, and halving the pitch will drop the sound by one octave. Modifying the pitch also changes the playback speed, so a sound with pitch greater than 1 will play at a faster speed, and a sound with a pitch less than 1 will play at a slower speed.

The example sets the pitch of sound 3 to 1.25.

SPAR handle parameter value 

SPAR 8 1 0.001       
SPAR 8 2 100

This instruction can be used to set sound rendering parameters for the sound source specified by handle. Currently, the following parameters are available:

Parameter Value Description
Rolloff Factor 1

The rolloff factor is used to control the rate at which sounds attenuate with respect to their distance away from the listener.

A higher rolloff factor will result in sounds becoming quieter faster, as the distance between the sound and the listener is increased. A rolloff of 0 will mean that sounds do not attenuate with respect to distance (the sounds will seem just as loud no matter how far away they are from the listener). For a client application using distance units of millimeters (so that position coordinates are often in magnitude of thousands), a rolloff factor of 0.001 is likely going to be appropriate. See the attenuation section for more information.

Reference Distance 2

The reference distance describes the radius for a sphere around a sound source in which the gain is not attenuated based on distance. If the distance between the sound and the listener is less than or equal to the reference distance, the sound's gain will not be attenuated (reduced) based on how far the sound is from the listener. Values must be greater than or equal to 0. See the attenuation section for more information.



The first example sets the rolloff factor of sound source 8 to 0.001, which is appropriate for generic sounds and positional units of millimeters.

The second example sets the reference distance of sound source 8 to 100.


Modifying Properties of the Listener

Message and Example(s) Description

GAIN value 

GAIN 0.0

Change the gain (volume) for the listener to value. The default gain for the listener is 1.0. This can be useful especially to mute all sounds for the listener, as the example demonstrates.

SLPO x y z 

SLPO 20.1 0.08 -9.23

Set the listener's position to <x, y, z>, in the same coordinate system as the sound sources. The default position is <0, 0, 0>

The example sets the position to <20.1, 0.08, -9.23>.

SLOR aX aY aZ uX uY uZ 

SLOR 1 0 0 0 1 0

Set the listener's orientation using two vectors. The first, <aX, aY, aZ>, specifies the "look-at" direction. The second, <uX, uY, uZ>, specifies the "up" direction. If two vectors are linearly dependent, the behavior is undefined. They do not need to be normalized. The defaults are <0, 0, -1> for the look-at vector and <0, 1, 0> for the up vector.

The example sets the listener's orientation to <1, 0, 0> for the look-at vector and <0, 1, 0> for the up vector.

SLVE x y z 

SLVE -2.0 0 0

Set the listener's velocity to <x, y, z>. Similar to setting a sound source's velocity, the server does not update the position of the listener based on this specified velocity. It is only used as a parameter for doppler effect calculations. See the doppler effect section for more information.

The example sets the velocity of the listener to -2.0 in the X direction.

Changing Global Parameters of Sound Rendering

Message and Example(s) Description

PARA parameter value 

PARA 1 343300       
PARA 3 0.001

This instruction can be used to set various global sound rendering parameters. Currently, the following parameters are available:

Parameter Value Description
Speed of Sound 1

The speed of sound is used for doppler effect calculations, and must be in the same units as the velocities of listener and sound sources. The default value is 343.3 meters per second, appropriate for the speed of sound through air.

Doppler Factor 2

The doppler factor can be used to emphasize or de-emphasize the pitch shifts due to the doppler effect. The default value is 1.0, and a value of 0 will effectively disable any pitch shifts due to the doppler effect.

Default Rolloff Factor 3

The default rolloff factor is used to control the rate at which sounds attenuate with respect to their distance away from the listener. NOTE: This property is not applied to existing sources - it only affects sources generated in the future. This parameter is provided so that the client can conveniently set the scale of the world during the initialization of their program.

A higher rolloff factor will result in sounds becoming quieter faster, as the distance between the sound and the listener is increased. A rolloff of 0 will mean that sounds do not attenuate with respect to distance (the sounds will seem just as loud no matter how far away they are from the listener). For a client application using distance units of millimeters (so that position coordinates are often in magnitude of thousands), a rolloff factor of 0.001 is likely going to be appropriate. See the attenuation section for more information.

Default Reference Distance 4

The reference distance describes a sphere around a sound source in which the gain is not attenuated based on distance. If the distance between the sound and the listener is less than or equal to the reference distance, the sound's gain will not be attenuated (reduced) based on how far the sound is from the listener. Values must be greater than or equal to 0. See the attenuation section for more information.

NOTE: This property is not applied to existing sources - it only affects sources generated in the future. This parameter is provided so that the client can conveniently configure the scale of the world during the initialization of their program.



The first example sets the speed of sound to 343300, corresponding to the units of millimeters per second, through the medium of air.

The second example sets the default rolloff factor to 0.001, which is appropriate for generic sounds and positional units of millimeters.

Deprecated Messages

Message and Example(s) Description

SSVE handle speed 

SSVE 6 3.5

(deprecated)
Set the sound's speed. The sound's current direction is used to compute the effective velocity. This assumes that the sound is moving in the same direction it is facing. Note that if the sound has no direction associated with it, the behavior is undefined. This message is leftover to support old applications that were tailored to the old Windows AudioServer project. New applications should avoid using this functionality, or use at their own risk.

The example sets sound 6's speed to 3.5.

Unsupported Messages

There were some messages that had inconsistent, incomplete, or altogether broken support in the old Windows AudioServer. This meant that in order to maintain full backwards compatibility, OAS would have had to implement these deprecated messages in a similar, broken manner. Instead, support for these messages has been dropped completely, under the assumption that old client applications would not be dependent on something that was broken. Support may be added in the future as needed.

Messages that are no longer supported: 

SSDR handle angle
SSRV handle angle gain
SSRV handle x y z gain

To-Do

Future Work

  • Enhance sockets code to add support for multiple connected clients. Asynchronous socket I/O.

Participants

Software Developers:

Project Advisors:

Misc. Development Assistance:

  • Philip Weber
  • Andrew Prudhomme


Initial Base Concept:

Retrieved from "http://ivl.calit2.net/wiki/index.php?title=OpenAL_Audio_Server&oldid=5005"