User`s guide
To Wave Device
5-446
the hardware throughput rate is higher than the simulation throughput rate, 
and the buffer tends to empty over the duration of the simulation.
Under normal operation, an empty buffer indicates that the simulation is 
finished, and the entire length of the audio signal has been processed. 
However, if the buffer size is too small in relation to the simulation throughput 
rate, the buffer may also empty before the entire length of signal is processed. 
This usually results in a device error or undesired device output. 
When the device fails to process the entire signal length because the buffer 
prematurely empties, you can choose to either increase the buffer size or the 
simulation throughput rate.
•Increase the buffer size. The 
Queue duration parameter specifies the length 
of signal, T
b
 (in real-time seconds), to buffer to the audio device during the 
simulation. The number of frames buffered is approximately
where F
s
 is the sample rate of the signal and M
o
 is the number of samples 
per frame. The optimal buffer size for a given signal depends on the signal 
length, the frame size, and the speed of the simulation. The maximum 
number of frames that can be buffered is 1024.
•Increase the simulation throughput rate. Two useful methods for improving 
simulation throughput rates are increasing the signal frame size and 
compiling the simulation into native code.
- Increase frame sizes (and convert sample-based signals to frame-based 
signals) throughout the model to reduce the amount of block-to-block 
communication overhead. This can drastically increase throughput rates 
in many cases. However, larger frame sizes generally result in greater 
model latency due to initial buffering operations. (Note that increasing the 
audio signal frame size does not affect the number of samples buffered to 
the hardware since the 
Queue duration is specified in seconds.)
- Generate executable code with Real-Time Workshop. Native code runs 
much faster than Simulink, and should provide rates adequate for 
real-time audio processing.
Audio problems at startup can often be corrected by entering a larger value for 
the 
Initial output delay parameter, which allows a greater portion of the 
T
b
F
s
M
o
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