

    <chapter id="rendering">
     <title>Listener and Sources</title>

    <sect1 id="object-state">
     <title>Basic Listener and Source Attributes</title>
    <para>
       This section introduces basic attributes which can be set both for 
       the Listener object and for Source objects.
    </para>

    <![ %RFC [
     <note id="rfc-bk000619-02"><title>RFC: attribute grouping</title><para>
       JM: "These attributes are of
       two types:  non-positional and positional.
       Non-positional properties include gain control and Environment Name."
     </para><para>
      I said: (low pass) Filters are applied to the sound during processing 
      at various stages. The exact sequence in which Filters are applied is 
      determined based on the location of the Objects they are 
      set for - spatial arrangement of Objects determines the
      sequence unless invariance is guaranteed, or invariance
     violation is permitted by the specification and current &AL;
     configuration state.
      </para><para>
      Is there a required order of application, i.e. a pipeline? 
      </para><para>
      Filter Parameters vs. Non-positional properties.
      Spatialization vs. Positional properties.
      Spatial attributes?
      Let's postpone grouping of attributes. 
      </para></note>
     ]]>

   
    <para>
      The &AL; Listener and Sources have attributes to describe 
      their position, velocity and orientation in three dimensional space.
      &AL; like &OGL;, uses a right-handed Cartesian coordinate system (RHS), 
      where in a frontal default view X (thumb) points right,
      Y (index finger) points up , and Z (middle finger) points towards 
      the viewer/camera. To switch from a left handed coordinate system (LHS)
      to a right handed coordinate systems, flip the sign on the Z coordinate.
    </para>

    <para>
    <table>
    <title>Listener/Source Position</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>POSITION</>
       <entry>3fv, 3f</>
       <entry> any except NaN </>
       <entry> { 0.0f, 0.0f, 0.0f } </>        
    </row>
    </tbody>
    </tgroup>
    </table>
    Description:
        POSITION specifies the current location of the Object in the
 	world coordinate system. Any 3-tuple of valid float/double values 
 	is allowed. Implementation behavior on encountering &NaN; and &Infty; 
        is not defined. The Object position is always defined in the
        world coordinate system.
    </para>

    <![ %Annote [
      <note><title>Annotation (No Transformation)</title><para>
        &AL; does not support transformation operations on Objects.
        Support for transformation matrices is not planned.
       </para></note>
    ]]>


    <para>
    <table>
    <title>Listener/Source Velocity</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>VELOCITY</>
       <entry>3fv, 3f</>
       <entry> any except NaN </>
       <entry> { 0.0f, 0.0f, 0.0f } </>        
    </row>
    </tbody>
    </tgroup>
    </table>     
    Description:
        VELOCITY specifies the current velocity (speed and direction) of
 	the Object, in the world coordinate system. Any 3-tuple of valid 
 	float/double values is allowed. The Object VELOCITY does not affect 
        its position.
        &AL; does not calculate the velocity from subsequent position
 	updates, nor does it adjust the position over time based on
 	the specified velocity. Any such calculation is left to the
 	application. For the purposes of sound processing, position and
 	velocity are independent parameters affecting different aspects
 	of the sounds. 
       </para><para>
        VELOCITY is taken into account by the driver to synthesize the 
        Doppler effect perceived by the Listener for each source, based 
        on the velocity of both Source and Listener, and the Doppler
        related parameters.
    </para>




    <para>
    <table>
    <title>Listener/Source Gain (logarithmic)</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>GAIN</>
       <entry>f</>
       <entry>0.0f, (0.0f, any</>
       <entry> 1.0f</>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description:
        GAIN defines a scalar amplitude multiplier. As a Source attribute, it applies
 	to that particular source only. As a Listener attribute, it effectively
        applies to all Sources in the current Context. The default 1.0 means 
        that the sound is un-attenuated. A GAIN value of 0.5 is equivalent to 
        an attenuation of 6 dB. The value zero equals silence (no output). Driver 
        implementations are free to optimize this case and skip mixing and 
        processing stages where applicable. The implementation is in charge of 
        ensuring artifact-free (click-free) changes of gain values and is free 
        to defer actual modification of the sound samples, within the limits of 
        acceptable latencies.
    </para>
    <para>
        GAIN larger than 1 (amplification) is permitted for Source and
        Listener. However, the implementation is free to clamp the
        total gain (effective gain per source times listener gain)
        to 1 to prevent overflow.
    </para>


    <![ %Annote [
      <note><title>Annotation/ Effective Minimal Distance</title><para>
         Presuming that the sample uses the entire dynamic range of
         the encoding format, an effective gain of 1 represents the 
         maximum volume at which a source can reasonably be played.
         During processing, the implementation combines the Source
         GAIN (or MIN_GAIN, if set and larger) with distance based
         attenuation. The distance at which the effective gain is 1
         is equivalent to the DirectSound3D MIN_DISTANCE parameter.  
         Once the effective gain has reached the maximum possible
         value, it will not increase with decreasing distance anymore.
      </para></note>
    ]]>

   
    <![ %Annote [
      <note><title>Annotation (Muting a Context)</title><para>
         To mute the current context, simply set Listener GAIN to zero.
         The implementation is expected to optimize for this case,
         calculating necessary (offset) updates but bypassing the 
         mixing and releasing hardware resources.
         The specification does not guarantee that the implementation
         will release hardware resources used by a muted context.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (Muting a Source)</title><para>
        To mute a Source, set Source GAIN to zero. The &AL; implementation
        is encouraged to optimize for this case. 
       </para></note>
    ]]>

    <![ %RFC [
    <para>
        <note id="rf-bk000503-01"><title>RFC: GAIN &gt; 1?</title><para>       
        GAIN could exceed 1 (to compensate attenuation elsewhere, or to account 
        for grater dynamic range of the hardware? No guarantees are made with 
        respect to range overflows? Precision loss? Culling by effective gain?
        Does &AL; clip values during processing, and when/at what stages?
        </para></note>

        <note id="rfc-bk000619-01"><title>RFC: Doppler</title><para>
        JM wrote: "VELOCITY is used by the driver
        to synthesize the Doppler effect perceived by the listener for each
 	source, based on the relative velocity of this source with respect
 	to the listener."
        Doppler is calculated using Source and Listener velocities measured 
        with respect to the medium. Do we have to account for the medium
        to move (offsetting listener/source) in later revisions (air/water currents)?
        </para></note>
   
        <note id="rfc-bk000619-03"><title>RFC: </title><para>
        JM removed: "For the purposes of sound processing, position and
        velocity are independent parameters affecting different paths
        in the sound synthesis." I think the "different aspects of sounds"
        is ambiguous. Is there a problem with describing &AL; as a
        multichannel processing machine?
        </para></note>
    </para>
     ]]>

    </sect1>


    <sect1 id="object-listener">
     <title>Listener Object</title>

    <para>
     The Listener Object defines various properties that affect processing of
     the sound for the actual output. The Listener is unique for an &AL; Context,
     and has no Name. By controlling the listener, the application controls
     the way the user experiences the virtual world, as the listener defines
     the sampling/pickup point and orientation, and other parameters that
     affect the output stream.
    </para>
    <para>
      It is entirely up to the driver and hardware configuration, i.e.
      the installation of &AL; as part of the operating system and
      hardware setup, whether the output stream is generated for
      headphones or 2 speakers, 4.1 speakers, or other arrangements,
      whether (and which) HRTF's are applied, etc..
    </para>
    
    <![ %Annote [
      <note><title>Annotation (Listener Anatomy)</title><para>
       The API is ignorant with respect to the real world 
       listener, it does not need to make assumptions on the 
       listening capabilities of the user, its species or its 
       number of ears. It only describes a scene and the position 
       of the listener in this scene.  It is the &AL; implementation
       that is designed for humans with ears on either side of the 
       head.
       </para></note>
    ]]>

 
    <![ %Annote [
      <note><title>Annotation (Listener State Evaluation)</title><para>
       Some Listener state (GAIN) affects only the very last
       stage of sound synthesis, and is thus applied to the sound stream 
       as sampled at the Listener position. Other Listener state is
       applied earlier. One example is Listener velocity as used to 
       compute the amount of Doppler pitch-shifting applied to each source:
       In a typical implementation, pitch-shifting (sample-rate conversion) 
       might be the first stage of the audio processing for each source.
       </para></note>
    ]]>


    <sect2>
    <title>Listener Attributes</title>

    <para>
      Several Source attributes also apply to Listener: e.g. POSITION, VELOCITY,
      GAIN. In addition, some attributes are listener specific.
    <table>
    <title>Listener Orientation</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>ORIENTATION</>
       <entry> fv </>
       <entry> any except NaN </>
       <entry>  { { 0.0f, 1.0f, 0.0f },       { 0.0f, 0.0f, -1.0f } } </>        
    </row>
    </tbody>
    </tgroup>
    </table>
      Description:
        ORIENTATION is a pair of 3-tuples representing the 'up' direction vector  
        and 'at' direction of the Object in Cartesian space. &AL; expects two
        vectors that are orthogonal to each other. These
        vectors are not expected to be normalized. If one or more vectors 
        have zero length, implementation behavior is undefined. If the two
        vectors are linearly dependent, behavior is undefined.
     </para>
     <![ %RFC [
      <note id="rfc-bk000503-01"><title>RFC: Orientation Paranoia</title><para>
        Watch LHS vs. RHS (sign on 'at'). 
        Confirm sequence is (up, at) not vice versa.
        Do we want to allow for different representations down the road?
      </para></note>
     ]]>
    </sect2>

    <sect2>
    <title>Changing Listener Attributes</title>
    <para>
      Listener attributes are changed using the Listener group of commands.
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> Listener{n}{sifd}{v} </function></funcdef>
      <paramdef> &enum; <parameter> paramName </parameter></paramdef>
      <paramdef> &type; <parameter> values </parameter></paramdef>
      </funcprototype></funcsynopsis>
    </para>
    </sect2>

    <sect2>
    <title>Querying Listener Attributes</title>
    <para>
      Listener state is maintained inside the &AL; implementation and can be
      queried in full. See Querying Object Attributes. The valid values for 
      paramName are identical to the ones for the Listener* command.
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> GetListener{sifd}v </function></funcdef>
      <paramdef> &enum; <parameter> param </parameter></paramdef>
      <paramdef> &type;* <parameter> values </parameter></paramdef>
      </funcprototype></funcsynopsis>
    </para>
    </sect2>
   </sect1>



  <sect1 id="object-sources">
     <title>Source Objects</title>
     <para>
       Sources specify attributes like position, velocity, and a buffer with
       sample data. By controlling a Source's attributes the
       application can modify and parameterize the static sample data provided
       by the Buffer referenced by the Source.
       Sources define a localized sound, and encapsulate 
       a set of attributes applied to a sound at its origin, i.e. in the 
       very first stage of the processing on the way to the listener. 
       Source related effects have to be applied 
       before Listener related effects unless the output is invariant
       to any collapse or reversal of order. 
     </para>
     <para>
      &AL; also provides additional functions to manipulate and query the
      execution state of Sources: the current playing status of a 
      source (started, stopped, paused), including access to the current
      sampling position within the associated Buffer. 
     </para>
  
    <![ %RFC [
      <note id="rfc-briareos000629-01"><title>RFC: Mike on Source Types</title><para>  
        AL_SOURCE_ABSOLUTE and AL_SOURCE_AMBIENT have been
        deprecated. AL_SOURCE_ABSOLUTE was simply the converse of the
        AL_SOURCE_RELATIVE pname, and as such was unnecessary. The
        effect of AL_SOURCE_AMBIENT is trivially emulated by either
        querying the Listener position and setting the Source position
        accordingly, or setting the Source position to (0,0,0) and the
        type to AL_SOURCE_RELATIVE, and is therefore also unnecessary.
      </para></note>
   ]]>
 

   <![ %RFC [
     <note id="rfc-bk000721-02"><title>RFC: Bernd on Source Types</title><para>
       Mike seems to miss a few problems here. W/o a converse we can't
       reset source attributes to ABSOLUTE. Ambient sounds are not 
       necessarily trivial. A3D manual suggested some magic number
       to fake the effect of ambient (i.e. sound that ideally
       can't be localized by listener). If we can get away with such magic 
       numbers in a tutorial in a driver-independent way, fine. If there is any
       risk that the impression of ambient sound requires driver specific
       hacks, then we need AMBIENT. As soon as we have a third localization
       type, ABSOLUTE and RELATIVE are needed as there is no unambiguous
       converse.

       From the A3D 2.0 Optimize.doc: 
       "Adding some ambient background noise is a great way to fill in the gaps 
        when the audio content is reduced.  A great way to make an ambient sound 
        seem like it is coming from everywhere is to load up two buffers with the 
        same sound, and position them about 2 meters behind the listener at 
        about 4 and 8 o\rquote clock.  The waves have to be looping (make sure
        there is no beating when you play them back).  Starting the sounds 180 
        degrees out of phase can help, as will playing them with slightly different 
        pitch-shift values." 
     </para></note>
   ]]>

   <![ %RFC [
     <note id="rfc-bk000721-03"><title>RFC: Bernd on Source Types (2)</title><para>
       There is a point to be made in using POSITION_RELATIVE and
       VELOCITY_RELATIVE iff we do not have AMBIENT to consider.
       This makes it a call-by-call choice when setting Source3f{v} 
       vectors, as it is applied when dereferencing.
   </para></note>
   ]]>

   <![ %RFC [
     <note id="rfc-bk000721-04"><title>RFC: Bernd on Source Types (3)</title><para>
       Semantically, AMBIENT has nothing to do with coordinate systems, 
       it is a qualifier just like multichannel direct passthru.
   </para></note>
   ]]>

    <![ %RFC [
      <note id="rfc-bk000721-05"><title>RFC:  Source Attenuation Clamping</title><para>  
         Using AL_SOURCE_ATTENUATION_MIN and AL_SOURCE_ATTENUATION_MAX
         to specify the clamping values for the normalized attenuation
         factor (which is a function of distance) is in contradiction
         to the distance based model that Creative is pushing for
         (DirectSound). As driver-interall culling of source and other
         processing might be based on the effective (overall, ultimate)
         gain composed of amplifications and attenuations accumulated
         over the entire processing, I raise the question whether a sound
         designer might not want to control the effective GAIN ranges
         instead of the distance attenuation itself. Samples commonly
         use the entire dynamic range provided by the format, which is
         mapped to the entire dynamic range of the output device. An
         effective gain exceeding 1 does not make sense, an amplification
         during processing might.
      </para></note>
    ]]>
    




    <sect2>
    <title>Managing Source Names</title>
    <para> 
     &AL; provides calls to request and release Source Names handles.
     Calls to control Source Execution State are also provided.
    </para>

    <sect3>
    <title>Requesting a Source Name</title>
    <para>
     The application requests a number of Sources using GenSources.
      <funcsynopsis><funcprototype> 
      <funcdef> &sizei; <function> GenSources </function></funcdef>
      <paramdef> &sizei; <parameter> n </parameter></paramdef>
      <paramdef> &uint;* <parameter> sources </parameter></paramdef>
      </funcprototype></funcsynopsis>
    </para>
   </sect3>


   <sect3>
   <title>Releasing Source Names</title>
   <para>
     The application requests deletion of a number of Sources
     by DeleteSources.
      <funcsynopsis><funcprototype> 
      <funcdef> &void; <function> DeleteSources </function></funcdef>
      <paramdef> &sizei; <parameter> n </parameter></paramdef>
      <paramdef> &uint;* <parameter> sources </parameter></paramdef>
      </funcprototype></funcsynopsis>
    </para>
   </sect3>

   <sect3>
   <title>Validating a Source Name</title>
   <para>
     The application can verify whether a source name is valid
     using the IsSource query.
      <funcsynopsis><funcprototype> 
      <funcdef> &bool; <function> IsSource </function></funcdef>
      <paramdef> &uint; <parameter> sourceName </parameter></paramdef>
      </funcprototype></funcsynopsis>
    </para>
   </sect3>
   </sect2>


    <sect2>
    <title>Source Attributes</title>
    <para>
        This section lists the attributes that are set per Source,
        affecting the processing of the current buffer. Some of
        these attributes can also be set for buffer queue entries.
    </para> 
    <![ %Annote [
      <note><title>Annotation (No Priorities)</title><para>
         There are no per Source priorities, and no explicit priority
         handling, defined at this point. A mechanism that lets the
         application express preferences in case that the implementation
         provides culling and prioritization mechanisms might be added
         at some later time. This topic is under discussion for GL as
         well, which already has one exlicit priority API along with 
         internally used MRU heuristics (for resident texture memory). 
       </para></note>
    ]]>
 



    <sect3>
    <title>Source Positioning</title>
    <para>
    <table>
    <title>SOURCE_RELATIVE Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry> SOURCE_RELATIVE </>
       <entry> &bool; </>
       <entry> FALSE, TRUE</>
       <entry> FALSE </>        
    </row>
    </tbody>
    </tgroup>
    </table>
       SOURCE_RELATIVE set to TRUE indicates that the values
       specified by POSITION are to be interpreted relative 
       to the listener position.
    </para>


    <![ %Annote [
      <note><title>Annotation (Position only)</title><para>
        SOURCE_RELATIVE does not affect velocity or orientation
        calculation.
       </para></note>
    ]]>

    </sect3>

    <sect3>
    <title>Buffer Looping</title>
    <para>
    <table>
    <title>Source LOOPING  Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <colspec colname=c1>
    <colspec colname=c2>
    <colspec colname=c3>
    <colspec colname=c4>
    <spanspec spanname=hspan namest=c1 nameend=c4 align=left>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry> LOOPING </>
       <entry> &uint; </>
       <entry> TURE, FALSE</>
       <entry> FALSE </>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description:
       LOOPING is a flag that indicates that the Source will not 
       be in STOPPED state once it reaches the end of last buffer
       in the buffer queue. Instead, the Source will immediately
       promote to INITIAL and PLAYING. The default value is FALSE.
       LOOPING can be changed on a Source in any execution state.
       In particular, it can be changed on a PLAYING Source.
    </para>

    <![ %Annote [
      <note><title>Annotation (Finite Repetition)</title><para>
        Finite reptition is implemented by buffer queueing.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (Loop Control)</title><para>
        To implement a 3 stage "loop point" solution, the
        application has to queue the FadeIn buffer first,
        then queue the buffer it wants to loop, and set 
        LOOPING to TRUE once the FadeIn buffer has been
        processed and unqueued. To fade from looping, the 
        application can queue a FadeOut buffer, then
        set LOOPING to false on the PLAYING source. Alternatively,
        the application can decide to not use the LOOPING
        attribute at all, and just continue to queue the buffer
        it wants repeated. 
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (Rejected alternatives)</title><para>
        A finite loop counter was rejected because it is
        ambiguous with respect to persistent (initial counter) 
        vs. transient (current counter). For similar reasons,
        a Play-equivalent command with a (transient) loop counter
        was rejected.  
      </para></note>
    ]]>
    </sect3>

    <sect3>
    <title>Current Buffer</title>
    <para>
    <table>
    <title>Source BUFFER Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>BUFFER</>
       <entry> ui </>
       <entry> any valid bufferName </>
       <entry> &NONE; </>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description: 
        Specify the current Buffer object, which means the 
        head entry in its queue.  Using BUFFER with the
        Source command on a STOPPED or INITIAL Source 
        empties the entire queue, then appends 
        the one Buffer specified.
    </para>
    <para>
       For a PLAYING or PAUSED Source, using the Source command 
       with BUFFER is an INVALID_OPERATION. 
       It can be applied to INITIAL and STOPPED Sources only.
       Specifying an invalid bufferName will
       result in an INVALID_VALUE error while specifying an
       invalid sourceName results in an INVALID_NAME error.
    </para>
    <para>
        NONE, i.e. 0, is a valid buffer Name.
        Source( sName, BUFFER, 0 ) is a legal way to release the
        current buffer queue on an INITIAL or STOPPED Source,
        whether it has just one entry (current buffer) or more.
        The Source( sName, BUFFER, NONE) call still causes an 
        INVALID_OPERATION for any source PLAYING or PAUSED, 
        consequently it can not be abused to mute or stop a source.
    </para>

    <![ %Annote [
      <note><title>Annotation (repeated Source+BUFFER does not queue) </title><para>
            Using repeated Source(BUFFER) calls to queue a buffer on
            an active source would imply that there is no way to
            release the current buffer e.g. by setting it to 0.
            On the other hand read-only queues do not allow for
            releasing a buffer without releasing the entire queue.

            We can not require BUFFER state to be transient and lost 
            as soon as a Source is implicitely or explicitely stopped. 
            This contradicts queue state being part of the Source's 
            configuration state that is preserved through Stop() 
            operations and available for Play().
       </para></note>
    ]]>

  
    </sect3>


    <sect3>
    <title>Queue State Queries</title>
    <para>
    <table>
    <title>BUFFERS_QUEUED Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry> BUFFERS_QUEUED </>
       <entry> &uint; </>
       <entry> [0, any]</>
       <entry> none </>        
    </row>
    </tbody>
    </tgroup>
    </table>
       Query only. Query the number of buffers in the queue
       of a given Source. This includes those not yet played,
       the one currently playing, and the ones that have been 
       played already. This will return 0 if the current and 
       only bufferName is 0.
    </para>


    <para>
    <table>
    <title>BUFFERS_PROCESSED Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry> BUFFERS_PROCESSED </>
       <entry> &uint; </>
       <entry> [0, any]</>
       <entry> none </>        
    </row>
    </tbody>
    </tgroup>
    </table>
       Query only. Query the number of buffers that have
       been played by a given Source. 
       Indirectly, this gives the index of the buffer
       currently playing. Used to determine how much
       slots are needed for unqueueing them.
       On an STOPPED Source, all buffers are processed.
       On an INITIAL Source, no buffers are processed,
       all buffers are pending.
       This will return 0 if the current and 
       only bufferName is 0.
    </para>

    <![ %Annote [
      <note><title>Annotation (per-Source vs. Buffer State)</title><para>
       BUFFERS_PROCESSED is only defined within the scope of a given
       Source's queue. It indicates that the given number of buffer names
       can be unqueued for this Source. It does not guarantee that the
       buffers can safely be deleted or refilled, as they might still be
       queued with other Sources. One way to keep track of this is to
       store, per buffer, the Source for which a given buffer was most
       recently scheduled (this will not work if Sources sharing buffers
       might be paused by the application). If necessary an explicit
       query for a given buffer name can be added in later revisions.
      </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (No Looping Queues)</title><para>
         Unqueueing requires nonzero BUFFERS_PROCESSED,
         which necessitates no looping on entire queues,
         unless we accept that no unqueueing is possible
         from Source looping over the entire queue.
         Currently not supported, as queueing is 
         primarily meant for streaming which implies
         unqueue-refill-requeue operations.
      </para></note>
    ]]>


    </sect3>

    <sect3>
    <title>Bounds on Gain</title>
    <para>
    <table>
    <title>Source Minimal Gain</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>MIN_GAIN</>
       <entry>f</>
       <entry>0.0f, (0.0f, 1.0f]</>
       <entry>0.0f</>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description:
        MIN_GAIN is a scalar amplitude threshold. It indicates the minimal GAIN 
        which is always guaranteed for this Source. At the end of the processing
        of various attenuation factors such as distance based attenuation and
        Source GAIN, the effective gain calculated is compared to this value.
        If the effective gain is lower than MIN_GAIN, MIN_GAIN is applied.
        This happens before the Listener GAIN is applied. If a zero MIN_GAIN 
        is set, then the effective gain will not be corrected.
    </para>


    <![ %Annote [
      <note><title>Annotation (Effective Maximal Distance)</title><para>
         By setting MIN_GAIN, the application implicitely defines a
         maximum distance for a given distance attenuation model and
         Source GAIN. The distance at which the effective gain is MIN_GAIN
         can be used as a replacement to the DirectSound3D MAX_DISTANCE parameter.  
         Once the effective gain has reached the MIN_GAIN value, it will 
         no longer decrease with increasing distance.
      </para></note>
    ]]>

   
    <para>
    <table>
    <title>Source Maximal Gain (logarithmic)</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>MAX_GAIN</>
       <entry>f</>
       <entry>0.0f, (0.0f, 1.0f]</>
       <entry>1.0f</>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description:
        MAX_GAIN defines a scalar amplitude threshold. It indicates the maximal 
        GAIN permitted for this Source. At the end of the processing
        of various attenuation factors such as distance based attenuation and
        Source GAIN, the effective gain calculated is compared to this value.
        If the effective gain is higher than MAX_GAIN, MAX_GAIN is applied.
        This happens before the Listener GAIN is applied. If the Listener gain
        times MAX_GAIN still exceeds the maximum gain the implementation can
        handle, the implementation is free to clamp. If a zero MAX_GAIN 
        is set, then the Source is effectively muted. The implementation is free
        to optimize for this situation, but no optimization is required or
        recommended as setting GAIN to zero is the proper way to mute a Source.
    </para>

    <![ %Annote [
      <note><title>Annotation (Un-attenuated Source)</title><para>
         Setting MIN_GAIN and MAX_GAIN to the GAIN value will effectively
         make the Source amplitude independent of distance. The 
         implementation is free to optimize for this situation. However, the
         recommended way to accomplish this effect is using a ROLLOFF_FACTOR 
         of zero.
       </para></note>
    ]]>



    <![ %Annote [
      <note><title>Annotation (Internal GAIN threshold)</title><para>
        The &AL; implementation is free to use an internally chosen 
        threshold level below which a Source is ignored for mixing. 
        Reasonable choices would set this threshold low enough so 
        that the user will not perceive a difference.  Setting MIN_GAIN
        for a source will override any implementation defined test.     
       </para></note>
    ]]>
    </sect3>



    <sect3>
    <title>Distance Model Attributes</title>
    <para>
    <table>
    <title> REFERENCE_DISTANCE Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry> REFERENCE_DISTANCE </>
       <entry> &float; </>
       <entry> [0, any]</>
       <entry> 1.0f </>        
    </row>
    </tbody>
    </tgroup>
    </table>
       This is used for distance attenuation calculations
       based on inverse distance with rolloff. Depending
       on the distance model it will also act as a distance
       threshold below which gain is clamped. See the
       section on distance models for details.
    </para>


    <para>
    <table>
    <title> ROLLOFF_FACTOR Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry> ROLLOFF_FACTOR </>
       <entry> &float; </>
       <entry> [0, any]</>
       <entry> 1.0f </>        
    </row>
    </tbody>
    </tgroup>
    </table>
       This is used for distance attenuation calculations
       based on inverse distance with rolloff. For
       distances smaller than MAX_DISTANCE (and, depending
       on the distance model, larger than REFERENCE_DISTANCE), 
       this will scale the distance attenuation over the
       applicable range. See section on distance models for
       details how the attenuation is computed as a function
       of the distance.
    </para>   
    <para>
       In particular, ROLLOFF_FACTOR can be set to zero for
       those Sources which are supposed to be exempt from
       distance attenuation. The implementation is encouraged
       to optimize this case, bypassing distance attenuation
       calculation entirely on a per-Source basis.
    </para>

    <para>
    <table>
    <title> MAX_DISTANCE Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry> MAX_DISTANCE </>
       <entry> &float; </>
       <entry> [0, any]</>
       <entry> MAX_FLOAT </>        
    </row>
    </tbody>
    </tgroup>
    </table>
       This is used for distance attenuation calculations
       based on inverse distance with rolloff, if the
       Inverse Clamped Distance Model is used. In this case,
       distances greater than MAX_DISTANCE will
       be clamped MAX_DISTANCE.
       MAX_DISTANCE based clamping is applied before MIN_GAIN clamping,
       so if the effective gain at MAX_DISTANCE is larger than MIN_GAIN,
       MIN_GAIN will have no effect. No culling is supported.     
    </para>

    <![ %Annote [
      <note><title>Annotation (No Culling)</title><para>
      This is a per-Source attribute supported for DS3D compatibility
      only. Other API features might suffer from side effects due to 
      the clamping of distance (instead of e.g. clamping to an effective
      gain at MAX_DISTANCE). 
      </para></note>
    ]]>

    </sect3>




    <sect3>
    <title>Frequency Shift by Pitch</title>
    <para>
    <table>
    <title>Source PITCH Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>PITCH</>
       <entry>f</>
       <entry> (0.0f, 1.0f]</>
       <entry> 1.0f</>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description: 
        Desired pitch shift, where 1.0 equals identity. Each reduction by 50 percent 
        equals a pitch shift of -12 semitones (one octave reduction). Zero is not
        a legal value.
    </para>
    </sect3>


    <sect3>
    <title>Direction and Cone</title>
    <para>
       Each Source can be directional, depending on the settings for
       CONE_INNER_ANGLE and CONE_OUTER_ANGLE. There are three zones
       defined: the inner cone, the outside zone, and the transitional
       zone in between. 
       The angle-dependent gain for a directional source is constant 
       inside the inner cone, and changes over the transitional zone
       to the value specified outside the outer cone.
       Source GAIN is applied for the inner cone,
       with an application selectable CONE_OUTER_GAIN factor to
       define the gain in the outer zone. In the transitional
       zone implementation-dependent interpolation between
       GAIN and GAIN times CONE_OUTER_GAIN is applied.


    </para>
    <![ %Annote [
      <note><title>Annotation (Interpolation Restrictions)</title><para>
         The specification does not specify the exact interpolation
         applied in the transitional zone, to calculate gain as a 
         function of angle. The implementation is free to use
         linear or other interpolation, as long as the values
         are monotonically decreasing from GAIN to GAIN times CONE_OUTER_GAIN.
       </para></note>
    ]]>


    <para>
    <table>
    <title>Source DIRECTION Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>DIRECTION</>
       <entry>3fv, 3f</>
       <entry> any except NaN </>
       <entry> { 0.0f, 0.0f, 0.0f } </>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description: 
        If DIRECTION does not equal the zero vector, the Source is directional. 
        The sound emission is presumed to be symmetric
        around the direction vector (cylinder symmetry). Sources are not
        oriented in full 3 degrees of freedom, only two angles are effectively
        needed.
       </para><para>
        The zero vector is default, indicating that a Source is not directional.
        Specifying a non-zero vector will make the Source directional. 
        Specifying a zero vector for a directional Source will effectively
        mark it as nondirectional. 
    </para>

    <![ %RFC [
       <note id="rfc-bk000821-01"><title>RFC: Oriented Sources </title><para>  
         Do we want an alternative AZIMUTH/ALTITUDE parametrization?
         Do we need ORIENTATION later? Is this superimposable? Can we mix both?
      </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (All Sources Directional)</title><para>
         From the point of view of the &AL; implementation, all
         Sources are directional. Certain choices for cone angles
         as well as a direction vector with zero length are treated
         equivalent to an omnidirectional source. The &AL; 
         implementation is free to flag and optimize these cases. 
       </para></note>
    ]]>

    <![ %RFC [
       <note id="rfc-bk000803-05"><title>RFC: Separate GenDirectionSource?</title><para>  
         Is there any risk that directional sources require different
         resources that have to be allocated from the beginning, and
         that we can not change an omnidirectional source to a
         bidirectional source at runtime?
      </para></note>
    ]]>


    <para>
    <table>
    <title>Source CONE_INNER_ANGLE Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>CONE_INNER_ANGLE</>
       <entry>i,f</>
       <entry>any except NaN</>
       <entry>360.0f</>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description: 
       Inside angle of the sound cone, in degrees. The default of 360 means that the
       inner angle covers the entire world, which is equivalent to an omnidirectional 
       source.
    </para>
    <![ %RFC [
       <note id="rfc-bk000926-01"><title>RFC: inconsistent cone angles? </title><para>  
        Is (inner &lt;= outer) required? Do we generate an error?
        Shouldn't this be a CONE_ANGLES 2f call specifying both angles at once?      
      </para></note>
    ]]>




    <para>
    <table>
    <title>Source CONE_OUTER_ANGLE Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>CONE_OUTER_ANGLE</>
       <entry>i,f</>
       <entry>any except NaN</>
       <entry>360.0f</>        
    </row>
    </tbody>
    </tgroup>
    </table>
     Description: Outer angle of the sound cone, in degrees. The default of 360 means that the
       outer angle covers the entire world. If the inner angle is also 360, then
       the zone for angle-dependent attenuation is zero.
    </para>
  
    <![ %RFC [
       <note id="rfc-bk000926-02"><title>RFC: addition? </title><para>  
     More generally, we could specify: 
      "If the sum of inner and outer angles is larger than 360,
      CONE_OUTER_ANGLE is clamped to (360-CONE_INNER_ANGLE) and
      there is no transition zone."
      </para></note>
    ]]>

    <para>
    <table>
    <title>Source CONE_OUTER_GAIN Attribute</title>
    <tgroup cols="4" align="left" colsep=1 rowsep=1>
    <thead>
    <row>
       <entry>&Par;</>
       <entry>&Sig;</>
       <entry>&Val</>
       <entry>&Def;</>        
    </row>
    </thead>
    <tbody>
    <row>
       <entry>CONE_OUTER_GAIN</>
       <entry>i,f</>
       <entry>[0.0f, 1.0f]</>
       <entry>0.0f</>        
    </row>
    </tbody>
    </tgroup>
    </table>
        Description: the factor with which GAIN is multiplied to 
        determine the effective gain outside the cone defined by
        the outer angle. The effective gain applied outside the
        outer cone is GAIN times CONE_OUTER_GAIN. Changing
        GAIN affects all directions, i.e. the source is attenuated 
        in all directions, for any position of the listener.
        The application has to change CONE_OUTER_GAIN as well if
        a different behavior is desired.
    </para>
   
  
    <![ %Annote [
      <note><title>Annotation (GAIN calculation)</title><para>
         The angle-dependend gain DGAIN is multiplied with the
         gain determined by the source's GAIN and any distance
         attenuation as applicable. Let theta be the angle
         between the source's direction vector, and the vector
         connection the source and the listener. This multiplier 
         DGAIN is calculated as:
        <literallayout>
          OUTER = CONE_OUTER_ANGLE/2;
          INNER = CONE_INNER_ANGLE/2;
          if      ( theta less/equal INNER )
              DGAIN = 1               
          else if ( theta greater/equal OUTER )
             DGAIN = CONE_OUTER_GAIN
          else
             DGAIN = 1 - (1-CONE_OUTER_GAIN)*((theta-INNER)/(OUTER-INNER))
          GAIN *= DGAIN
        </literallayout>
         in the case of linear interpolation. The implementation
         is free to use a different interplation across the (INNER,OUTER)
         range as long as it is monotone. 
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (CONE_OUTER_GAIN always less than GAIN)</title><para>
         CONE_OUTER_GAIN is not an absolute value, but (like all GAIN
         parameters) a scaling factor. This avoids a possible error 
         case (implementations can count on effective gain outside the
         outer cone being smaller than GAIN), and ensures the common
         case in which changing GAIN should affect inner, transitional, 
         and outer zone simultaneously.
      </para><para>
         In case that the application desires to have an outer zone
         volume exceeding that of the inner cone, the mapping to
         &AL; will require to rotate the Source direction to the
         opposite direction (negate vector), and swapping
         inner and outer angle.
      </para></note>
    ]]>
    </sect3>
   </sect2>


    <sect2>
    <title>Changing Source Attributes</title>
    <para>
      The Source specifies the position and other properties as
      taken into account during sound processing. 
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> Source{n}{sifd} </function></funcdef>
      <paramdef> &uint; <parameter> sourceName </parameter></paramdef>
      <paramdef> &enum; <parameter> paramName </parameter></paramdef>
      <paramdef> &type; <parameter> value </parameter></paramdef>
      </funcprototype></funcsynopsis>
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> Source{n}{sifd}v </function></funcdef>
      <paramdef> &uint; <parameter> sourceName </parameter></paramdef>
      <paramdef> &enum; <parameter> paramName </parameter></paramdef>
      <paramdef> &type;* <parameter> values </parameter></paramdef>
      </funcprototype></funcsynopsis>
    </para>
   </sect2>

   <sect2>
    <title>Querying Source Attributes</title>
    <para>
      Source state is maintained inside the &AL; implementation, and the
      current attributes can be queried. The performance of such queries is 
      implementation dependent, no performance guarantees are made. The 
      valid values for the paramName parameter are identical to the ones 
      for Source*.
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> GetSource{n}{sifd}{v} </function></funcdef>
      <paramdef> &uint; <parameter> sourceName </parameter></paramdef>
      <paramdef> &enum; <parameter> paramName </parameter></paramdef>
      <paramdef> &type;* <parameter> values </parameter></paramdef>
      </funcprototype></funcsynopsis>
    </para>

    <![ %Scratch [
     <warning><para>
      Old signature: T GetSource{sifd}{v}( uint id, enum param );
     </para></warning>
     ]]>
 
    </sect2>



  <sect2 id="queueing">
   <title>Queueing Buffers with a Source</title>
   <para>
     &AL; does not specify a built-in streaming mechanism. There
     is no mechanism to stream data e.g. into a Buffer object.
     Instead, the API introduces a more flexible and versatile
     mechanism to queue Buffers for Sources.
   </para>
   <para>
     There are many ways to use this feature, with
     streaming being only one of them. 
    <itemizedlist>
    <listitem>
    <para>
      Streaming is replaced by queuing static
      buffers. This effectively moves any multi-buffer
      caching into the application and allows the
      application to select how many buffers it wants
      to use, whether these are re-used in cycle,
      pooled, or thrown away. 
   </para>
   </listitem>
   <listitem>
   <para>
      Looping (over a finite number of repititions) can be 
      implemented by explicitely repeating buffers
      in the queue. Infinite loops can (theoretically)
      be accomplished by sufficiently large repetition counters.
      If only a single buffer is supposed to be repeated
      infinitely, using the respective Source attribute is
      recommended.
   </para>
   </listitem>
   <listitem>
   <para>
      Loop Points for restricted looping inside a buffer
      can in many cases be replaced by splitting the 
      sample into several buffers, queueing the sample
      fragments (including repetitions) accordingly.
   </para>
   </listitem>
   </itemizedlist>
   Buffers can be queued, unqueued after they have been
   used, and either be deleted, or refilled and queued again.
   Splitting large samples over several buffers maintained
   in a queue has a distinct advantages over approaches that
   require explicit management of samples and sample indices.
   </para>

    <![ %RFC [
      <note id="bk000626-01"><title>RFC: Unified Handling</title><para>  
       Jonathan Blow has proposed removing the distinction between
       streaming and non-streaming buffers. An existing example is
       the unified for directional and omnidirectional sources, where
       all sources are treated as directional.
      </para></note>
    ]]>


   <sect3>
   <title>Queueing command</title>
   <para>
     The application can queue up one or multiple buffer names 
     using SourceQueueBuffers. The buffers will be queued in the sequence
     in which they appear in the array.
      <funcsynopsis><funcprototype> 
      <funcdef> &void; <function> alSourceQueueBuffers </function></funcdef>
      <paramdef> &uint; <parameter> sourceName </parameter></paramdef>
      <paramdef> &sizei; <parameter> numBuffers </parameter></paramdef>
      <paramdef> &uint; * <parameter> bufferNames </parameter></paramdef>
      </funcprototype></funcsynopsis>
     This command is legal on a Source in any state (to allow for
     streaming, queueing has to be possible on a PLAYING Source).
     Queues are read-only with exception of the unqueue operation. 
     The Buffer Name NONE (i.e. 0) can be queued.
    </para>
  
    <![ %Annote [
      <note><title>Annotation (BUFFER vs. SourceQueueBuffers)</title><para>
       A Sourcei( sname, BUFFER, bname ) command is an immediate
       command, and executed immediately. It effectively unqueues
       all buffers, and then adds the specified buffer to the
       then empty queue as its single entry. Consequently, this
       call is only legal if SourceUnqueueBuffers is legal.
       In particular, the Source has to be STOPPED or INITIAL.
       The application is still obliged to delete all
       buffers as were contained in the queue.  
       Sourcei( sname, BUFFER, NONE ) is a legal command,
       effectively wiping the queue without specifying an
       actually playable buffer.
       </para></note>
    ]]>


    <![ %Annote [
      <note><title>Annotation (Buffer Repetition)</title><para>
        To accomplish a finite number of repetitions of a buffer name multiple times,
        the buffer has to be queued multiple times. If the need occurs, the
        API could be extended by SourceQueueBuffer( sname, bname, repetitions )
        call for brevity.
      </para></note>
    ]]>

    <![ %RFC [
       <note id="rfc-bk000806-04"><title>RFC: Duration of bName==0? </title><para>  
          The buffer is considered empty, it should have zero length,
          thus zero duration for consistency. If an application wants to
          schedule a pause, specifying duration for a gain==0 queue entry
          might be a cleaner solution.     
      </para></note>
    ]]>


    <![ %Annote [
      <note><title>Annotation (Backwards Compatiblity)</title><para>
         Sourcei( sname, BUFFER, bname ) has been rejected as
         a queueing command, as it would make semantics dependent on
         source state (queueing if PLAYING, immediate else).
         The command is not legal on a PLAYING or PAUSED Source.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (No BUFFER_QUEUE)</title><para>
          Duplication of one entry point is preferable to
          duplicating token enums, and tokens do not express
          commands, but specify the attribute/state affected.
          From the same reason, there is no BUFFER_UNQUEUE
          token-as-command.
       </para></note>
    ]]>

    </sect3>

   <sect3>
   <title>Unqueueing command</title>
   <para>
     Once a queue entry for a buffer has been appended to a queue 
     and is pending processing, it should not be changed.
     Removal of a given queue entry is not possible unless 
     either the Source is STOPPED (in which case then entire queue
     is considered processed), or if the queue entry has already
     been processed (PLAYING or PAUSED Source).
   </para>
   <para>
     The Unqueue command removes a number of buffers entries that 
     have finished processing, in the order of appearance, from 
     the queue. The operation will fail if more buffers are
     requested than available, leaving the destination arguments 
     unchanged. An INVALID_VALUE error will be thrown.  
     If no error, the destination argument will have been updated 
     accordingly.
      <funcsynopsis><funcprototype> 
      <funcdef> void <function>  SourceUnqueueBuffers </function></funcdef>
      <paramdef> &uint;    <parameter> sourceName </parameter></paramdef>
      <paramdef> &sizei;    <parameter> numEntries </parameter></paramdef>
      <paramdef> &uint;*    <parameter> bufferNames </parameter></paramdef>
      </funcprototype></funcsynopsis>
    </para>

    <![ %Annote [
      <note><title>Annotation (Unqueueing shared buffers)</title><para>
         If a buffer is queued with more than one source, it might have
         been processed for some not all of them. With the current
         interface, the application is forced to maintain its own list
         of consumers (Sources) for a buffer it wishes to unqueue.
         For groups of Sources that are never individually PAUSED 
         nor STOPPED, the application can save the MRU Source for 
         which the buffer was scheduled last.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (Looping a Queue vs. Unqueue):</title><para>
          If a Source is playing repeatedly, it will traverse
          the entire Queue repeatedly. Consequently, no buffer
          in the queue can be considered processed until 
          there is no further repetition scheduled.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (No Name based access)</title><para>
         No interface is provided to access a queue entry by name,
         due to ambiguity (same buffer name scheduled several times
         in a sequence).
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (No Index based access)</title><para>
         No interface is provided for random access to a queue entry 
         by index.
       </para></note>
    ]]>
    </sect3>



   <![ %Annote [
     <sect3>
     <title>More Annotation on Queueing</title>

    <![ %Annote [
      <note><title>Annotation (No Queue Copying)</title><para>
          The current queue of a source could be copied to another source, 
         as repetition and traversal parameters are stored unless the 
         queue entry is unqueued, or the queue is replaced using 
         AL_BUFFER.  Copying a queue is a special case of
         copying Source state in one sense, and a special case of
         a synching problem in another. Due to these unresolved issues
         no such command is included in the current specification.
         To share queues, the application can keep buffer names
         and the selected attributes that define the queue entries
         in an array or other lookup table.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (No Explicit QueueClear)</title><para>
        Sourcei( sname, BUFFER, NONE ) serves the 
        same purpose. The operation is also redundant
        with respect to Unqueue for a STOPPED Source.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (Queueing vs. AppendData):</title><para>
         Buffer queueing does not solve the synchronization and timing 
         issues raised by possible underflow, as these are inherent
         to application-driven (pushed) streaming. However, it turns 
         an internal AL error condition (offset exceeds valid data)
         into an audible artifact (Source stops).
         Its main advantage is that it allows the application coder
         to operate at a scale of her own choice, selecting the
         number and size of buffers used for caching the stream,
         and to schedule buffer refill and queueing according to 
         preferences and constraints. Queueing effectively moves 
         all problems related to replacing or appending Buffer data 
         to the scale of entire arrays istead of single samples and 
         indices.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (Multiple Sources on a stream)</title><para>
         Queueing allows for the application to determine how much of
         a backlog of the data stream is preserved. The application can
         keep buffers, and queue them with other Sources after they have
         been used already by the original Source. Unlike the mechanism
         for appending data to a buffer, the backlog is visible to the
         application and under its control, and no synchronization of
         Sources using the stream is required.
       </para></note>
    ]]>



    <![ %Annote [
      <note><title>Annotation (Loop Points and Compressed Data)</title><para>
        For compressed data, uncompression by the application might be
         impossible or undesireable. In consequence, splitting the sample
         into several buffers is not possible without explicit support
         by the API. Buffer-Buffer operations will be added as needed,
         for the time being applications should not try to use compressed
         samples if more than full looping is required.
       </para></note>
    ]]>



    <![ %Annote [
      <note><title>Annotation (No Explicit Queue Objects)</title><para>
        Explicit Queue objects have been considered and rejected,
        as they introduce another producer-consumer dependency with
        another level of indirection. Further, e.g. QUEUE would 
        also require deprecating BUFFER (breaking backwards 
        compatibility) as an alSource argument, or would introduce
        a confusing set of precedence and overide rules if both
        are used in sequence. However, in the absence of explicit
        queue objects the application will be forced to keep track
        where buffers have been queued in case it intends to 
        unqueue them for refill or deletion. If several sources
        use the same buffers (e.g. for synchronous or 
        asynchronous streaming) the buffer will have to be
        unqueued from each single one.
      </para></note>
    ]]>


    <![ %Scratch [
      <note><title>Annotation (Queue no Display List)</title><para>
         An interface resembling &OGL; display-lists has been 
         considered and rejected. The problem with this approach 
         is that not only commands would have to be prohibited 
         (similarly, not all GL calls are legal within a display 
         list), but also parameters (enumerations).

         In particular, only a small set of operations is meant
         to be legal for a queue at this point, and appending
         to a queue has to be possible at any time.
         Within a hypothetical AL display list, only relative
         timing/conditionals are allowed as arguments. This
         might necessitate to have multiple forms for deferred
         commands, or to not allow for absolute timing.

         Example:
         <literallayout>
         // lock this queue for overwriting/appending
         alBeginQueue( qname, APPEND | REPLACE );  

         // queue a buffer in sequence, with parameters
         // boolean: never skip? never bail?
         alQueue( AL_BUFFER, bid, loopdir, repetitions );
         ...

         // end lock. 
         // Existing  queue content will be replaced
         //  or appended at this point.
         alEndQueue();
       </literallayout>
       </para></note>
    ]]>


    <![ %Example [
      <example>
      <title>Queue Then Delete </title>
      <programlisting>

         create source
         queue buffer1
         queue buffer2
         queue buffer3
         play
         request deletion of buffer1,2,3
       
       </programlisting>
       </example>
    ]]>

    <![ %Example [
      <example>
      <title> Queue and Refill with Dual Buffering</title>
      <programlisting>

        create source
        fill buffer1
        queue buffer1
        play
        fill buffer2
        queue buffer2
        check for unused buffers
          unqueue buffer1
          fill buffer1
          queue buffer1
          ...
       
       </programlisting>
       </example>
    ]]>

    <![ %Example [
      <example>
      <title> Queue for Loop Points</title>
      <programlisting>

         create source
         read sample data
         split sample data into pre/main/post
         queue pre
         queue main with repetitions
         queue post
         play
         set repetitions to 0 on main when needed
         wait till post has been played
       
       </programlisting>
       </example>
    ]]>
 
   </sect3>
]]>

<![ %Scratch [
  <sect3>
    <title>Attributes Specific to Queueing</title>

     <section>
    <title>Buffer Traversal</title>
    <para>
       The Buffer traversal attribute specifies the direction
       in which the sample in the buffer is supposed to be
       processed. To account for the 3 basic modes of traversal that
       can be implemented in software and hardware, the following
       tokens are defined:
       <literallayout>
         LOOP_DIRECTION     /* traversal direction */

         FORWARD            /* linear forward   (increment) */
         BACKWARD           /* linear backward  (decrement) */
         FORWARD_AND_BACK   /* RESERVED: ping-pong-looping  */
 
       </literallayout>
       The first and the next two tokens are legal with a buffer queue command. 
       They are not legal for a Source command, in any possible
       Source state. The last token is reserved, but not yet legal to use.
    </para>  


    <![ %Annote [
      <note><title>Annotation (Ping-Pong postponed)</title><para>
        Definition and implementation of ping-pong looping
        has been postponed. Applications can fake it at doubling
        memory expense by reverse copying the buffer (2nd buffer queued 
        or in a double size single buffer). If there is hardware support
        for this feature, AL will have to support it eventually. A boolean
        flag is not acceptable because of this possibility.
       </para></note>
    ]]>
    </sect3> 
]]> <!-- SCRATCH -->
 
  
   </sect2>
 





    <sect2>
    <title>Managing Source Execution</title>
    <para>
       The execution state of a source can be queried. &AL; provides
       a set of functions that initiate state transitions causing
       Sources to start and stop execution.
    </para>
    <para>
       TBA: State Transition Diagram.
    </para>

    <![ %Annote [
      <note><title>Annotation/ Source Config/Exec State</title><para>
       Sources have configuration state and execution state.
       Configuration state is directly set by the application using 
       AL commands, starting with the INITIAL configuration. Execution 
       state (e.g. the offset to the current sample) is not under direct 
       application control and not exposed. 
       </para></note>
    ]]>

    <sect3>
    <title>Source State Query</title>
    <para>
      The application can query the current state of any Source 
      using GetSource with the parameter Name SOURCE_STATE. 
      Each Source can be in one of four possible execution states: 
      INITIAL, PLAYING, PAUSED, STOPPED. Sources that are either
      PLAYING or PAUSED are considered active. Sources that are
      STOPPED or INITIAL are considered inactive. Only PLAYING
      Sources are included in the processing. The implementation 
      is free to skip those processing stages for Sources that 
      have no effect on the output (e.g. mixing for a Source 
      muted by zero GAIN, but not sample offset increments). 
      Depending on the current state of a Source certain (e.g. repeated)
      state transition commands are legal NOPs: they will be ignored, 
      no error is generated.
    </para>
    </sect3>

    <sect3>
    <title>State Transition Commands</title>
    <para>
      The default state of any Source is INITIAL. From this state
      it can be propagated to any other state by appropriate use
      of the commands below. There are no irreversible state
      transitions. 
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> SourcePlay </function></funcdef>
      <paramdef> &uint; <parameter> sName </parameter></paramdef>
      </funcprototype></funcsynopsis>
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> SourcePause </function></funcdef>
      <paramdef> &uint; <parameter> sName </parameter></paramdef>
      </funcprototype></funcsynopsis>
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> SourceStop </function></funcdef>
      <paramdef> &uint; <parameter> sName </parameter></paramdef>
      </funcprototype></funcsynopsis>
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> SourceRewind </function></funcdef>
      <paramdef> &uint; <parameter> sName </parameter></paramdef>
      </funcprototype></funcsynopsis>


    </para>
    <para>
      The functions are also available as a vector variant,
      which guarantees synchronized operation on a set of 
      Sources.
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> SourcePlayv </function></funcdef>
      <paramdef> &sizei; <parameter> n </parameter></paramdef>
      <paramdef> &uint;* <parameter> sNames </parameter></paramdef>
      </funcprototype></funcsynopsis>
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> SourcePausev </function></funcdef>
      <paramdef> &sizei; <parameter> n </parameter></paramdef>
      <paramdef> &uint;* <parameter> sNames </parameter></paramdef>
      </funcprototype></funcsynopsis>
      <funcsynopsis><funcprototype> 
      <funcdef> void <function> SourceStopv </function></funcdef>
      <paramdef> &sizei; <parameter> n </parameter></paramdef>
      <paramdef> &uint;* <parameter> sNames </parameter></paramdef>
      </funcprototype></funcsynopsis>

      <funcsynopsis><funcprototype> 
      <funcdef> void <function> SourceRewindv </function></funcdef>
      <paramdef> &sizei; <parameter> n </parameter></paramdef>
      <paramdef> &uint;* <parameter> sNames </parameter></paramdef>
      </funcprototype></funcsynopsis>

    </para>
    <para>
      The following state/command/state transitions are defined:
      <itemizedlist>
       <listitem>
       <para>
        Play() applied to an INITIAL Source will promote the Source
        to PLAYING, thus the data found in the Buffer will be fed 
        into the processing, starting at the beginning.
        Play() applied to a PLAYING Source will restart the Source
        from the beginning. It will not affect the configuration,
        and will leave the Source in PLAYING state, but reset the
        sampling offset to the beginning.
        Play() applied to a PAUSED Source will 
        resume processing using the Source state 
        as preserved at the Pause() operation.
        Play() applied to a STOPPED Source will propagate it
        to INITIAL then to PLAYING immediately.
       </para> 
       </listitem>
       <listitem>
       <para>
        Pause() applied to an INITIAL Source is a legal NOP.
        Pause() applied to a PLAYING Source will change its state to
        PAUSED. The Source is exempt from processing, its current
        state is preserved.
        Pause() applied to a PAUSED Source is a legal NOP.
        Pause() applied to a STOPPED Source is a legal NOP.
       </para>
       </listitem>
       <listitem>
       <para>
        Stop() applied to an INITIAL Source is a legal NOP.
        Stop() applied to a PLAYING Source will change its state to
        STOPPED. The Source is exempt from processing, its current
        state is preserved.
        Stop() applied to a PAUSED Source will change its state
        to STOPPED, with the same consequences as on a PLAYING
        Source.
        Stop() applied to a STOPPED Source is a legal NOP.
       </para>
       </listitem>
       <listitem>
       <para>
        Rewind() applied to an INITIAL Source is a legal NOP.
        Rewind() applied to a PLAYING Source will change its state to
        STOPPED then INITIAL. The Source is exempt from processing, 
        its current state is preserved, with the exception of the 
        sampling offset which is reset to the beginning.
        Rewind() applied to a PAUSED Source will change its state
        to INITIAL, with the same consequences as on a PLAYING
        Source.
        Rewind() applied to a STOPPED Source promotes the Source
        to INITIAL, resetting the sampling offset to the beginning.
       </para>
       </listitem>
    </itemizedlist>
    </para>

    <![ %Annote [
      <note><title>Annotation (SourceNext)</title><para>
         The specification does not provide any means to
         immediately skip from the current Buffer to the
         next in the queue.  A conditional stop (following
         the next complete traversal) is available.
         If necessary an additonal entry point could be
          provided in future revisions.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (Rewind() optional)</title><para>
         The INITIAL state is not identical to the STOPPED state.
         Applications that want to verify whether a Source
         has indeed been PLAYING before becoming STOPPED can
         use Rewind() to reset the Source state to INITIAL.
         This is an optional operation that can safely be
         omitted by application without this constraint. 
         Applications that want to guard against Play() on
         a Source that is INITIAL can query the Source state
         first.         
      </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (Play() on a PLAYING Source)</title><para>
         Repeated Play() commands applied a PLAYING Source are
         interpreted as an (atomic) sequence to stop and restart a
         Source. This can be used by applications that want to restart
         a sound but do not care whether the Source has finished or not,
         and do not want an audible pause. One example is the DOOM 
         chaingun repeatedly abbreviating the pistol sound. To guard
         against redundant Play() commands, an application can query
         the current state before executing Play(). If the application
         coder wants to be sure that the Source will play the buffer
         again, she can either increment PLAY_COUNT, or queue the buffer.
       </para></note>
    ]]>

    <![ %Annote [
      <note><title>Annotation (redundant commands)</title><para>
         The simple variant (e.g. SourcePlay) is redundant to
         the vector variant (e.g. SourcePlayv). However, these
         calls will be used frequently, and the simple variant
         is provided for convenience. However, &AL; does not
         enable applications to use literals as source names.
       </para></note>
    ]]>

    </sect3>

    <sect3>
    <title>Resetting Configuration</title>
    <para>
      The INITIAL state is not necessarily identical to the 
      default state in which Source is created. INITIAL merely
      indicates that the Source can be executed using the
      SourcePlay command. A STOPPED or INITIAL Source can
      be reset into the default configuration by using a
      sequence Source commands as necessary. As the application
      has to specify all relevant state anyway to create a
      useful Source configuration, no reset command is provided.
    </para>
  

    <![ %RFC [
       <note id="rfc-bk000802-01"><title>RFC: remove INITIAL</title><para>  
         INITIAL is identical to STOPPED. The only additional information
         conveyed is that INITIAL indicates a source has never been played. 
         Once a Source is STOPPED, it is not possible by state query alone 
         to decide whether it has played again. If Sources are used only 
         once, an application can use INITIAL to verify a Source has been 
         played.
         The problem that I have with this is that if we acknowledge that
         the application might need to verify a Source has played once,
         why force the application to throw away Sources to accomplish
         this? An explicit state PLAYABLE replacing INITIAL and its
         inauspicious connotations (default state) and a state transition
         function Rewind() that makes a STOPPED Source PLAYABLE again would 
         be one possibility to address this need. The obvious drawback is
         that it breaks backwards compatibility.
      </para></note>
    ]]>


    <![ %RFC [
       <note id="rfc-bk000731-01"><title>RFC: state issues </title><para>  
        A Source is active if it is PAUSED or PLAYING.

        A Source that is STOPPED preserves configuration state,
          including buffer/queue information.

        Only a Source that is Reset() to INITIAL looses all
          buffer and queue information. In this case, the INITIAL

        Sources will be stopped implicitely when reaching the
           end of a non-repeating (non-looping) buffer traversal.
        Sources can be stopped explicitely by the application
        with either Stop() or Reset(). 

       Stop() propagates
        the source to STOPPED preserving its configuration state,
        setting its execution state to the same as if it reached
        the end of execution.
  
      </para></note>
    ]]>


    <![ %Annote [
      <note><title>Annotation (illegal NOPs)</title><para>
          In the current specification there are no illegal NOPs.
          In other words, no sequence of commands affecting the
          execution state will generate an INVALID_OPERATION error.  
      </para></note>
    ]]>


     <![ %RFC [
     <note><title>RFC/bk000504:</title><para>
       No  UNDEFINED state. Always valid state. I.e. we have a default Buffer
       that is used for sources where the application doesn't specify,
       and what's in it? Default gain is zero? We need to specify
       INITIAL.
     </para></note>

     <note><title>RFC/bk000504:</title><para>
       Potential ambiguity: how to we distinguish STOPPED as
       requested by the application from INACTIVE for
       non-looping sounds once the buffer has been iterated?
       Related: handling of Sources using an underflowing
       streaming buffer? IMO not recommended, make this
       undefined on error.
     </para></note>


     <note><title>RFC/bk000504:</title><para>
       Possible redundancy: the only reason for STOP seems to
       be resetting the play positions. Redundant if we
       ever manipulate offsets directly (rewind/set).
     </para></note>

     <note><title>RFC/bk000504:</title><para>
       Possible redundancy:
       If we ever want to support explicit setting of the start
       position/offset into Buffer, START is equivalent to Set(0).
       Also see LOOP (implies has to be STOPPED). Fade-Out and
       Fade-In control - always manually?
     </para></note>

     ]]>     
   </sect3>
   </sect2>
   </sect1>
   </chapter>


