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XStudio Voice Tracker

It is possible that a combination of factors comes together to cause what might be called a "hesitation" in playback of audio when rehearsing sequences. If there is a noticeable delay between the time you click on the [Play] button and when you hear audio - on the order of 1/2-second or more, there are some things that can be done to improve response time.

The problem most generally manifests itself when the playback audio folder is on a remote machine, as would be the case with centrally-stored audio.

Here are some steps to try to improve response time - try them in order, shutting down the audio engine before making the changes, then restarting the audio engine and testing the change. It is recommended you make one change at a time to get a good sense of the effect of each change rather than making several changes and not knowing which of them may have fixed the problem.

1.Change the audio stream buffer size. The default audio stream buffer size is 32768 (32,768 bytes). Open the registry, locate the key PlayBufferSize (see the Registry Information section for details on its location) and change the value to 8196 (8,196 bytes). Relaunch the audio engine and test the playback start response. If the response is acceptable, shut the audio engine down and increase the value to 16384 (16,384 bytes). Re-test playback response. If it's still acceptable, try increasing the value to 24580 (24,580 bytes) and repeat the test. If playback response time is acceptable, leave the setting. If not, revert the setting back to the last value you tested that produced acceptable results.

Lowering the PlayBufferSize value increases hard disk read frequency. The idea is to use the largest value possible while getting acceptable play start response time. This is particularly true of situations where the playback audio is stored on a central server - you want to keep disk access to a minimum to keep from "hammering" the central storage device any more than necessary.

2.Change the pipe clock setting.  If you've changed the buffer size down to 8196 and are still not getting acceptable results, you can try changing the pipe clock setting. Note changing this value may cause some audio cards to fail altogether, including some Digigram models.
 
Ensure the audio engine is shut down, then open the registry and locate the key PipeClock. See the Registry Information section for details on its location. Change the value from it's default of 44100 to 32000, restart the audio engine and test playback response. You can also try setting the value to 48000.
 
By way of explanation, if the source playback audio was sample at 32kHz and the PipeClock setting is 44100 (44.1kHz), the audio engine is rate-converting, which adds some time to the process. The idea is to match this value to the value of the majority of the source audio.

Changing the PipeClock value produces best results when you know all or most of the source playback audio was sampled at the same rate. For audio files produced on a DCS system, this would be 32000 (32kHz).