| Commit message (Collapse) | Author | Age | Files | Lines |
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* Add EAX extensions (EAX 2.0-5.0, X-RAM)
* Comment out C++17 leftovers
* Remove everything related to patching
* Update alsoftrc.sample
* Rewrite integration
* Fix GCC compilation under Linux
* Always reset EAX effect properties when loading it into FX slot
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This isn't great since it can fail when PipeWire is handling audio but no
devices are available at initialization, causing the Pulseaudio or ALSA backend
to be selected instead. Future versions of PipeWire are expected to have a
better way to detect if it's handling audio, but for now this is better than
nothing.
A config option is available for users to have the PipeWire backend be usable
even with no devices at initialization, just in case.
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This simply omits the scale factor from the filter, similar to how up-sampling
does. The consequence of this is less smooth transitions when ramping the
pitch while down-sampling, but otherwise behaves fine.
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Unfortuantely, the relevant KDE bug still seems to exist (streams are forced to
KDE's default device after opening, even when they're created with a specific
device at user request). I do not know why KDE thinks this is in any way good
behavior (the user doesn't get their desired device, nor does the stream get
the appropriate format for the device its ultimately put on), but making
streams non-movable as a workaround has been a thorn in the side of non-KDE
users for too long. C'mon KDE, it's been nearly (if not more than) 7 years now.
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AL_LOKI_quadriphonic is left alone since that is what the extension is called
and what code expects. All other instances have been fixed for consistency.
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Turns out the C version of the cubic resampler is just slightly faster than
even the SSE3 version of the FIR4 resampler. This is likely due to not using a
64KB random-access lookup table along with unaligned loads, both offseting the
gains from SSE.
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This improves the transition width, allowing more of the higher frequencies
remain audible. It would be preferrable to have an upper limit of 32 points
instead of 48, to reduce the overall table size and the CPU cost for down-
sampling.
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This improves a stereo (front-left + front-right) sound "image" by generating a
front-center channel signal. Done correctly, it helps reduce the comb effects
and phase errors associated with using only two speakers to simulate center
sounds.
Note that it shouldn't be used if the front-center channel is already included
in the positional audio mix (the dialog effect is okay). In general, it may
actually be better to exclude the front-center channel from the positional
audio mix and use this to generate front-center output.
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Expose it in alsoft-config as well.
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Also rename the 'paired' value to 'panpot', and make it the default.
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Instead of forcing the device to always use the specified send count, it simply
limits requests to it.
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The default number of auxiliary effect slots is now 64. This can still be
raised by the config file without a hard maximum, but incurs processing cost
for each effect slot generated by the app.
The default number of source sends is now actually 2, as per the EFX docs.
However, it can be raised up to 16 via ALC_MAX_AUXILIARY_SENDS attribute
requests, rather than the previous 4.
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It still requires a custom configuration to specify appropriate speaker
distances.
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Both 5.1 Side and Rear configurations use 'surround51' to look up the
appropriate decoder file. The decoder loader already handles mapping between
rear and side channels, so there's no need for separate options.
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Designed for apps that either don't change the listener's AL_GAIN, or don't
allow the listener's AL_GAIN to go above 1. This allows the volume to still be
increased further than such apps may allow, if users find it too quiet.
Be aware that increasing this can easily cause clipping. The gain limit
reported by AL_GAIN_LIMIT_SOFT is also affected by this.
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