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We had to lock audiocommon with the old asynchron HLE audio emulation,
now our Mixer is just a plain FIFO which may underrun.
Of course, this will stutter, but underruning the audio backend is likely worse.
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For more information:
https://docs.google.com/document/d/1tBEgsJh7QiwNwepXI0eobfK3U8LkJButSyeuFt1degM/edit?usp=sharing
removed: SSE includes (not used)
added: 16bit -> float -> 16bit functions
added: linear interpolator and high-quality (windowed-sinc) interpolator functions (including Resampler class)
added: dithering
changed: renamed variables and reformatted a few things to fit with style guide (braces, #include->const)
changed: use s16, u16, s32, u32 etc
changed: store samples and do all computations as floats
changed: volume from 0 - 255
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Emulated Wiimote speaker sounds will go to the host system's speakers. Real Wiimotes will continue to use their own speaker for Wiimote speaker sounds. All Wiimote speaker sound can be disabled by unchecking the "Enable Speaker Data" option.
Each emulated Wiimote can have its speaker routed from left to right via the "Speaker Pan" setting in the emulated wiimote settings dialog. Use any value from -127 for leftmost to 127 for rightmost with 0 being the centre.
Added code in the InputConfig to use a spin control for non-boolean values.
Defaulted the setting of "Enable Speaker Data" to disabled.
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The Wiimotes are positioned as follows:
Wiimote 0 = Center
Wiimote 1 = Left
Wiimote 2 = Right
Wiimote 3 = Center
The Wiimote speaker output can be disabled via the "Enable Speaker Data" checkbox in the Wiimote settings.
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Remove audio frame limit.
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It serves no purpose and causes bugs and confusion for users.
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The two instances of this class were sharing a frac variable causing
audio glitches when both were running (which is now all the time).
Fixes issue 7463 (Since DTK merge, audio has staic in it).
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The code actually handles this case correctly; the algorithm is linear
interpolation between the two closest samples, and the way it is written
should work correctly with any ratio.
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The primary motivation here is to make sure we submit samples from the
CPU thread. This makes sure the timing of related interrupts accurate,
and generally keeps the different kinds of audio synchronized. This will also
allow improvements to audio dumping functionality.
The new code is also more concise because it gets rid of some duplicated
audio mixing code.
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Atm this is hardcoded to '\t'.
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This commit intentionally breaks audio output from HLE Zelda UCode; no other functional change.
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Also moved && and || to ends of lines instead of start.
Fixed misc vertical alignments and some { needed newlining.
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OpenAL itself stretch the time on slowdowns, so the Mixer isn't allowed also to change the rate.
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The usual one is 32->48 khz interpolation. So there is no need in a special 1:1 interpolation only for performance.
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