User Manual

OPTIMIZING PERFORMANCE
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! Raising the buffer size to eliminate audio artefacts on playback is mainly effective if you are currently using
very small buffers, 64 to 256 samples. If the buffers are already big (1024 or 2048 samples) you will not notice
much difference.
Making Buffer Size adjustments in the Reason Preferences dialog
If you are running Reason under Windows and are using an ASIO driver, or under Mac OS X and are using a Core Au-
dio driver, you can adjust the input and output latencies in the Preferences – Audio dialog.
D This is done by dragging the Buffer Size slider.
General procedure for reducing latency
The basic procedure for optimizing the latency is the following:
1. Open a song and start playback.
You want to choose a song that is reasonably demanding, i.e. with more than just a few tracks and devices.
2. Open the Preferences dialog.
Under Mac OS X, this is found on the Reason menu; under Windows it’s found on the Edit menu.
3. Click the Audio tab and locate the Buffer Size slider.
! If you are making adjustments in the ASIO Control Panel for hardware with an ASIO driver (Windows only), you
should make a note of the current buffer settings before changing them.
4. While the song is playing, listen closely for pops and clicks and try lowering the latency (Buffer Size).
5. When you get pops and clicks, raise the Buffer Size value a bit.
6. Close the Preferences dialog (and ASIO Control Panel, if open).
About Latency Compensation
Below the Input Latency and Output Latency rows in the Preferences-Audio dialog, you will find an item called Re-
cording Latency Compensation. This value is used internally in Reason to compensate for latencies when recording
audio using external monitoring.
Recording Latency Compensation
! Adjusting the Recording Latency Compensation parameter is never necessary when you have selected “Auto-
matic” in the Monitoring section on the Audio page in Preferences - see “Monitoring”.
Recording Latency Compensation is when the program adjusts the position of the recorded audio according to the
current latency. Here's how it works:
If you're recording audio and are not monitoring through Reason (e.g. monitoring directly through the audio card or
via an external mixer), the audio you record will reach the program slightly late. This is because you play along with
background tracks or the metronome - and you hear these delayed by the output latency. Also, the sound you record
is sent to the program via the buffers in the audio hardware - it is delayed by the input latency.