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* Don't return whether the bsinc filter cuts or notChris Robinson2018-01-101-1/+1
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* Use one macro to handle both resample padding sizesChris Robinson2018-01-091-6/+5
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* Combine the chorus and flanger processing functionsChris Robinson2018-01-092-16/+7
| | | | | | | Given that they're nearly identical, it should be relatively simple to use the same effect state to process either of them, similar to the reverbs. The big differences seem to be the delay range (much shorter with flanger) and the defaults.
* Rename the device's temp buffer storage to be more genericChris Robinson2018-01-091-5/+2
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* Remove standard reverb specific processing functionsChris Robinson2018-01-071-1/+0
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* Move the UNEXPECTED macro to the main header and rename itChris Robinson2018-01-071-0/+8
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* Use a separate function to get the cubic valueChris Robinson2018-01-071-0/+9
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* Remove the sinc4 tableChris Robinson2018-01-071-5/+0
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* Replace the sinc4 resampler with cubicChris Robinson2018-01-072-6/+2
| | | | | | | 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.
* Make MixSamples non-static globalChris Robinson2017-12-171-1/+2
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* Allow storing multiple buffers in a ALbufferlistitemChris Robinson2017-12-152-68/+145
| | | | | | | | | | | | | | | This will be to allow buffer layering, multiple buffers of the same format and sample rate that are mixed together prior to resampling, filtering, and panning. This will allow composing sounds from individual components that can be swapped around on different invocations (e.g. layer SoundA and SoundB on one instance and SoundA and SoundC on a different instance for a slightly different sound, then just SoundA for a third instance, and so on). The longest buffer within the list item determines the length of the list item. More work needs to be done to fully support it, namely the ability to specity multiple buffers to layer for static and streaming sources. Also the behavior of loop points for layered static sources should be worked out. Should also consider allowing each layer to have a sample offset.
* Add queries to get the source offset with the device clockChris Robinson2017-12-032-0/+34
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* Add an "un-exposed" method to get the library versionChris Robinson2017-10-161-0/+10
| | | | | | | This reports the same ALSOFT version as alGetString(AL_VERSION), but doesn't require a current context (which requires a ALCdevice) to call. Do *NOT* use this version to determine feature support, use the standard interfaces. If you think you need to use this, you probably don't, and shouldn't.
* Avoid a separate function to query ambisonic mode supportChris Robinson2017-09-271-12/+8
| | | | | | Now FuMa and ACN channel orders are required, as are FuMa, SN3D, and N3D normalization schemes. An integer query (alcGetIntegerv) is added for the maximum ambisonic order.
* Re-update effect slots when context properties changeChris Robinson2017-09-278-48/+42
| | | | | Also keep all free property update structs together in the context instead of per-object.
* Don't update context and listener props unnecessarilyChris Robinson2017-09-274-40/+52
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* Update the context state properties separatelyChris Robinson2017-09-275-26/+69
| | | | | | | | | | | | | The context state properties are less likely to change compared to the listener state, and future changes may prefer more infrequent updates to the context state. Note that this puts the MetersPerUnit in as a context state, even though it's handled through the listener functions. Considering the infrequency that it's updated at (generally set just once for the context's lifetime), it makes more sense to put it there than with the more frequently updated listener properties. The aforementioned future changes would also prefer MetersPerUnit to not be updated unnecessarily.
* Add an option to ignore the app's speed of sound for reverb decayChris Robinson2017-09-222-1/+3
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* Pass the context to the auxiliary effect update methodChris Robinson2017-09-211-2/+2
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* Manually save and restore the FPU rounding mode on WindowsChris Robinson2017-09-191-0/+3
| | | | | | | Apparently there is a bug with at least MinGW-W64 where fegetenv and fesetenv do not properly save and restore the FPU rounding mode, resulting in the rounding mode remaining as round-to-zero after certain function calls. I do not know if this also affects MSVC, but better safe than sorry for now.
* Rename resampler labelsChris Robinson2017-08-271-3/+3
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* Add a higher quality bsinc resampler using 24 sample pointsChris Robinson2017-08-272-5/+12
| | | | | | | 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.
* Rename the bsinc resampler to bsinc12Chris Robinson2017-08-252-4/+4
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* 0 meters per unit is invalidChris Robinson2017-08-211-1/+1
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* Pass the filter entry to apply to resample_fir4Chris Robinson2017-08-181-3/+2
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* Store the sinc4 table in the filter stateChris Robinson2017-08-161-7/+11
| | | | Also rename the resampler functions to remove the unnecessary '32' token.
* Simplify bsinc filter storage in the filter stateChris Robinson2017-08-161-6/+5
| | | | | | | Rather than storing individual pointers to filter, scale delta, phase delta, and scale phase delta entries, per phase index, the new table layout makes it trivial to access the per-phase filter and delta entries given the base offset and coefficient count.
* Keep bsinc info together in a structChris Robinson2017-08-151-2/+1
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* Avoid re-selecting the direct HRTF mix functionChris Robinson2017-08-071-0/+2
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* Add a front-stablizer config option for surround sound modesChris Robinson2017-07-311-0/+3
| | | | | | | | | | | | 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.
* Apply the output buffer offset before writing to itChris Robinson2017-07-151-1/+1
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* Use macros to set and restore the mixer FPU modeChris Robinson2017-07-132-4/+14
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* Store the default effect slot in the contextChris Robinson2017-07-132-8/+5
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* Store the QSA backend's ExtraData in the wrapper structChris Robinson2017-06-291-2/+0
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* Remove the fastf2u conversion functionChris Robinson2017-06-271-5/+0
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* Clean up some messy rounding codeChris Robinson2017-06-261-0/+18
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* Use the bsinc resampler for the converterChris Robinson2017-06-251-0/+2
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* "Convert" the QSA backend to the new APIChris Robinson2017-06-181-19/+0
| | | | | | | | | | I say "convert" because it takes the lazy way and essentially just embeds the wrappers into the backend. It's done this way because I lack the means to check any changes, even syntactically. This also means the device's ExtraData field is still needed. However, this does mean all the backends are now using the new API. Code related to the old interface can now be removed.
* Make the dithering depth configurableChris Robinson2017-06-171-1/+1
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* Don't force a fade-in when resuming a paused sourceChris Robinson2017-06-091-11/+1
| | | | | | | | This needs to be handled more automatically by the mixer to work correctly. Otherwise, requiring a property update on resume can put the source into a playing state with the mixer never playing it, due to not having valid mixing parameters and the mixing parameters not getting calculated because no updates are specified by the app (and forcing an update can break deferred updates).
* Fix source sends' initial HF absorption and decay calculationChris Robinson2017-05-272-2/+2
| | | | | | | | The HF absorption is applied given the source distance, as relative to the source's immediate environment, with additional absorption being applied given the room/reverb environment. This does double up the amount of absorption compared to the dry path, but it can be assumed the initial reflections travel a longer distance.
* Add a new compressor/limiterChris Robinson2017-05-272-24/+30
| | | | | | This is just for the output limiter right now, but in the future can be used for the compressor EFX effect. The parameters are also hardcoded, but can be made configurable after 1.18.
* Finalize ALC_SOFT_output_limiterChris Robinson2017-05-251-5/+0
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* Add an option to dither 8- and 16-bit outputChris Robinson2017-05-232-0/+7
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* Reduce the amount of variables that hold the same valueChris Robinson2017-05-211-2/+2
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* Add a method to copy a filter's coefficientsChris Robinson2017-05-212-0/+10
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* Use a macro to specify the decay target gainChris Robinson2017-05-211-0/+3
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* Apply more proper air absorption to the wet pathChris Robinson2017-05-192-0/+4
| | | | | | | | This properly accounts for the room rolloff factor for normal air absorption (which makes it none by default, like distance attenuation), and uses the reverb's decay time, decay hf ratio, decay hf limit, and room air absorption properties to calculate an initial hf decay with the WetGainAuto flag. This mirrors the behavior of the initial distance decay.
* Finalize AL_SOFT_source_spatializeChris Robinson2017-05-111-6/+0
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* Reduce LIMITER_VALUE_MAXChris Robinson2017-05-091-1/+1
| | | | | | The previous value couldn't actually be expressed as a float and got rounded up to the next whole number value, leaving the potential for an overrun in the squared sum.
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