Would fmod be suitable for a digital audio workstation kind of application?

In your docs you say:
The mixer engine block size in samples. Use this to adjust mixer update granularity. Default = 1024. (milliseconds = 1024 at 48khz = 1024 / 48000 * 1000 = 21.33ms). This means the mixer updates every 21.33ms.

The mixer engine number of buffers used. Use this to adjust mixer latency. Default = 4. To get the total buffersize multiply the bufferlength by the numbuffers value. By default this would be 4*1024.

That is quite high for the pro audio world.
Is fmod capable of achieving the ultra low latency available from ASIO/WASAPI/CoreAudio devices (with sufficient CPU provided)?
There are cards that can do 0.7ms in ASIO and WASAPI can usually go down to 3.0ms with generic onboard sound.
Can numbuffers be set to 1?

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[quote:39l1p7in]Would fmod be suitable for a digital audio workstation kind of application?[/quote:39l1p7in]
It [i:39l1p7in]could[/i:39l1p7in] be used for that kind of application, it [i:39l1p7in]could[/i:39l1p7in] definately have much lower latency than the default values.

You can experiment with different values of bufferlength and numbuffers. Given sufficient CPU it can acheive very low latency. Taken from the docs:

[size=150:39l1p7in]Low latency recording support[/size:39l1p7in]
[size=125:39l1p7in]FMOD Ex now supports super low latency recording, processing and output through a new recording engine.
Via ASIO the recording->DSP->playback latency can be as low as 1-3ms! This is great for realtime processing and playback of recorded audio.[/size:39l1p7in]

Having said all that, I think it really depends on the scale of your project and the features you need. FMOD is very flexible and quite feature rich, but make sure it can does everything you need before you commit to it.

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