[META] Harmonize Pa_OpenStream() suggestedLatency param to native buffer size calculations

[PortAudio-admin]

Issue created by @RossBencina

Different host APIs interpret Pa_OpenStream suggestedLatency values differently. To some extent this is OK because it reflects the differences in buffering mechanisms. However, as much as possible we have decided that all host APIs should use similar calculations.

Close when #205 is closed. (see Resolution criteria below)

This is discussed in detail here: BufferingLatencyAndTimingImplementationGuidelines

Earlier discussion between Dmitry and Ross is archived here: http://music.columbia.edu/pipermail/portaudio/2010-July/010681.html

Ross is reviewing WMME, DS and ASIO with regard to this ticket.

The current ASIO computations may be incorrect (see ticket #93)

93 PA/ASIO !SelectHostBufferSize() doesn't take user framesPerBuffer into account

This ticket is also related to the way default latency values are computed (see ticket #97)

This ticket is related to the way suggestedLatency==0 is interpreted (ticket #99)

ALSA, !CoreAudio etc will need to be reviewed once further progress is made on this ticket.

182 PA/CoreAudio ignores Pa_OpenStream() suggestedLatency param

=== Resolution criteria ===

This ticket will be closed when the following two conditions are met:

• A review of native buffer size computation code has been undertaken, and the method used to compute host buffer sizes is concisely summarised on the wiki page for each host API.
• A visual inspection of the graphs created by patest_suggested_vs_streaminfo_latency appears consistent with our expectations (using WMME and DS as a guide).

Both of these tasks are part of code review ticket #205.

[PortAudio-admin]

Comment by @RossBencina

after a quick re-test with WMME it seems like it's better to use at least 3 buffers if possible (total latency with 2 buffers is higher than with 3 to get stable results).

I'm working on some code that tests different buffer values to see what works best.

here's a page that says KMixer uses 10ms buffers internally: http://msdn.microsoft.com/en-us/library/ff536484(VS.85).aspx

http://www.microsoft.com/whdc/device/audio/default.mspx

[PortAudio-admin]

Comment by @RossBencina

Dmitry says:

3. Computation of buffering/buffer sizes based on suggested latency. For my application WASAPI and Alsa operate ok.

[PortAudio-admin]

Comment by @RossBencina

r1744 mostly resolves this for DirectSound, however in the case of full-duplex streams with fixed userFramesPerBuffer, the buffer adapter adds extra output latency equal to userFramesPerBuffer -- this extra latency is not taken into account when interpreting suggestedLatency. Therefore full duplex output currently reports userFramesPerBuffer more latency than requested.

[PortAudio-admin]

Comment by @RossBencina

TRAC migration: propagating ticket status from TRAC