| Commit message (Collapse) | Author | Age | Files | Lines |
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This has a couple behavioral changes. First and biggest is that querying
AL_BUFFERS_PROCESSED from a source will always return all buffers processed
when in an AL_STOPPED state. Previously all buffers would be set as processed
when first becoming stopped, but newly queued buffers would *not* be indicated
as processed. That old behavior was not compliant with the spec, which
unequivocally states "On a source in the AL_STOPPED state, all buffers are
processed."
Secondly, querying AL_BUFFER on an AL_STREAMING source will now always return
0. Previously it would return the current "active" buffer in the queue, but
there's no basis for that in the spec.
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This avoids using seq_cst for loading the source state when either inside the
mixer, or otherwise protected from inconsistencies with async updates. It also
fixes potential race conditions with getting the source offset just as a source
stops.
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This change allows pair-wise panning to mostly go through the normal ambisonic
panning methods, with one special-case. First, a term is added to the stereo
decoder matrix's X coefficient so that a centered sound is reduced by -3dB on
each output channel. Panning in front creates a similar gain response to the
typical
L = sqrt(1-pan)
R = sqrt(pan)
for pan = [0,1]. Panning behind the listener can reduce (up to) an additional
-10dB, creating a audible difference between front and back sounds as if
simulating head obstruction.
Secondly, as a special-case, the source positions are warped when calculating
the ambisonic coefficients so that full left panning is reached at -30 degrees
and full right at +30 degrees. This is to retain the expected 60-degree stereo
width. This warping does not apply to B-Format buffer input, although it
otherwise has the same gain responses.
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Also rename the 'paired' value to 'panpot', and make it the default.
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Since we're now waiting for space to be available before mixing, the mixing
buffer isn't adding another period.
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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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The voices are still all allocated in one chunk to avoid memory fragmentation.
But they're accessed as an array of pointers since the size isn't static.
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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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Perf shows less than 1 percent CPU difference from the higher quality bsinc
resampler, but uses almost twice as much memory (a 128KB lookup table).
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A device will never have 0 latency. OpenAL Soft itself uses a sample buffer
length of UpdateSize*NumUpdates, and during playback will have about
(NumUpdates-1) periods filled, more or less. Without a more accurate
measurement from the playback system, this is better than reporting 0.
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This places the Send[] array at the end of the struct, making it easier to
handle dynamically.
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ALsourceProps' Send[] array is placed at the end of the struct, and given an
indeterminate size. Extra space is allocated at the end of each struct given
the number of auxiliary sends set for the device.
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Since it's modified by the mixer when playback is ended, a plain struct member
isn't safe.
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Also explicitly link with libz for alffplay, since static ffmpeg libs need it.
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This also removes the buffer queue callback's call to aluMixData, which could
potentially block on a mutex.
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Which can happen with native-only apps
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