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Set accumulated capture-latency for physical-outputs
and accumulated playback-latency for physical-inputs
after Ardour is done setting all non-physical port latencies.
This will be needed for latency-compensation of the complete graph.
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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
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class RingBuffer<> is a very generic name and should not pollute the
global namespace.
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pthread-w32 does not support pthread_setschedparam() with
SCHED_FIFO and bails out. While pthread_create() simply ignores the policy
and sets the priority regadless.
This only affects ctrl-surface event-loops & AutomationWatch on Windows.
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This hopefully fixes an issue with port-registration (new session)
being skipped because PortAudioBackend::available() still false
until the first callback.
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This fixes an issue with FaderPort8 (and maybe other surfaces or synths).
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The same device may have different names for input + output
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This issue is not always reproducible but when it does occur it happens
somewhat consistently on both 32bit and 64bit builds(Tested on Windows 7).
The midiOutReset call does not return (or it takes so long that it might as
well be indefinite) and as it is not strictly necessary just remove it.
Resolves: http://tracker.ardour.org/view.php?id=7095
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When failing to open audio stream in callback mode
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uselib is no longer implicit (inherited by .use). This is still incomplete,
some uselibs for non-linux variants may be missing.
bld.is_defined("HAVE_XXX") also no longer works and will have to be
changed (I think to bld.env["HAVE_XXX"]) in countless places.
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and callback API
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blocking API
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Apparently this is now required by my version of gcc/mingw
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portaudio backend
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this just works (no symbol conflicts)
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For drivers that correctly report latency values(ASIO) this should result a
much closer alignment of audio in a loopback test. Measurement and
adjustment may still be needed, especially for non-ASIO drivers.
Testing with the RME HDSP Multiface and Yamaha AG06 using ASIO drivers results
in maximum offset of a couple of samples.
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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.
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default
Don't use the callback API for now until further and wider testing.
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midiInReset triggers the sysex callback to tell the application that it has
finished with the buffer. Calling midiInAddBuffer results in an infinite loop
so just return during shutdown.
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So they can be used to give a contextual error message in the GUI
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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.
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