PortAudio uses what it calls 'default suggested latencies' but in callback streaming mode, they can result in wildly inaccurate buffer sizing (e.g. the user requests a buffer size of 128 but PortAudio actually instructs ASIO to use a much bigger size).
What we do now is to improve PortAudio's suggested latency calculation by basing it on the actual buffer size requested by the user.
Generated by tools/f2s. Some hand-editing will be required in a few places to fix up comments related to timecode
and video in order to keep the legible
This has been tested on four devices:
- A RME HDSP Multiface
- A Yamaha AG06
- A Focusrite 2i2
- A built-in soundcard running ASIO4ALL
The HDSP and the AG06 only return one buffer size when queried so the preferred
size is used as before.
The Focusrite returns a min corresponding to the position of the slider in the
control dialog and the max is 1024. The granularity is 1 so this means that the
number of values needs to be reduced for the current UI design with a combo
box so the granularity is increased until there are around 8-9 buffer sizes to
choose from evenly spaced between min and max(but we could easily change this
if the UI changes etc).
The ASIO4ALL driver returns a min of 64 and a max of 2048 and a granularity of
8. So where the minimum buffer size and granularity is a power of 2 use only
buffer sizes that are power of 2.
If the driver returns different values for min and max it is not currently
possible to indicate which is the driver preferred value. A checkbox or other
UI element could be added to the AudioSetup dialog to only use the preferred
value but that is more work and perhaps not necessary.
Having error codes defined in PortaudioIO means it is not dependent on the
ErrorCodes in AudioBackend but it doesn't really make sense to have another
set, so just use the PA ones until they become insufficient.
