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* Round the early and late delay tap sample offsetsChris Robinson2016-10-261-4/+4
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* Restore a comment that was accidentally deletedChris Robinson2016-10-101-0/+9
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* Better sort the main delay line tapsChris Robinson2016-10-061-60/+39
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* Decorrelate the early reflection inputsChris Robinson2016-10-061-10/+19
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* Pass current and target gains directly for mixingChris Robinson2016-10-051-45/+12
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* Update a comment about using row mixersChris Robinson2016-10-051-3/+1
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* Better pack the late reverb low- and all-pass variablesChris Robinson2016-10-051-21/+25
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* Use the row mixer functions for the B-to-A-Format conversionChris Robinson2016-10-041-16/+10
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* Enhance reverb using B-Format processingChris Robinson2016-10-031-405/+386
| | | | | | | | | | | | Technically it uses A-Format processing from the B-Format input and output. But this attempts to provide better spatial definition to the reverberation so that it can be used in a more generic fashion, allowing it to be decoded as any other B-Format signal to whatever output is needed, and also allowing for a bit of height information when the output is capable of such. There may still be some kinks to work out, such as properly decorrelating the early reflection taps and tweaking the late reverb density. But it seems to be a good enough start.
* Fix EAX reverb effect output for HRTF and UHJChris Robinson2016-09-131-2/+2
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* Call ALfilterState_processC directlyChris Robinson2016-09-122-5/+0
| | | | | | It's the only implementation currently, so there's no point to having it stored as a function pointer in the filter struct. Even if there were SIMD versions, it'd be a global selection, not per-instance.
* Write to the correct outputs for extra reverb channelsChris Robinson2016-09-111-2/+2
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* Combine the reverb decorrelator delay line with the main delay lineChris Robinson2016-09-111-46/+26
| | | | | | | Since it was merely acting as an extension of it anyway, with the second delay line tap (for late reverb) copying attenuated samples to the decorrelator line that was being tapped off of. Just extend the delay line and offset the decorrelator taps to be relative to the late reverb tap.
* Make the SelectMixer function sharableChris Robinson2016-09-061-15/+1
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* Use the optimized mixing functions for reverb outputChris Robinson2016-09-061-83/+117
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* Use deinterlaced buffers for the intermediate reverb storageChris Robinson2016-09-061-119/+137
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* Do reverb modulation before band-pass filteringChris Robinson2016-09-061-7/+6
| | | | | | | Ideally the band-pass should probably happen closer to output, like gain is. However, doing that would require 16 filters (4 early + 4 late channels, each with a low-pass and high-pass filter), compared to the two needed to do it on input.
* Do multiple samples at once for reverb modulationChris Robinson2016-09-051-38/+49
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* Use a predefined identity matrixChris Robinson2016-09-053-30/+6
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* Add a ref count to ALeffectStateChris Robinson2016-08-2510-203/+309
| | | | | This is mostly just reorganizing the effects to call the Construct method which initializes the ref count.
* Combine related members into a structChris Robinson2016-08-241-2/+2
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* Remove DevFmtBFormat3D, which is covered by DevFmtAmbi1Chris Robinson2016-07-311-1/+1
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* Simplify a format checkChris Robinson2016-07-301-3/+2
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* Add a config to output first-, second-, or third-order ambisonicsChris Robinson2016-07-291-1/+3
| | | | | | | | | Currently incomplete, as second- and third-order output will not correctly handle B-Format input buffers. A standalone up-sampler will be needed, similar to the high-quality decoder. Also, output is ACN ordering with SN3D normalization. A config option will eventually be provided to change this if desired.
* Remove the last use of ALfilterState_processSingleChris Robinson2016-07-261-3/+17
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* Remove broken autowah effect codeChris Robinson2016-07-261-271/+0
| | | | | It's been disabled forever, and I have no idea how to make it work properly. Better to just redo it when making something that works.
* Rename input_gain to b0Chris Robinson2016-07-263-6/+6
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* Avoid more uses of ALfilterState_processSingleChris Robinson2016-07-251-9/+9
| | | | | It's a horriobly inefficient way to process multiple samples through the filter.
* Avoid manual loops for ALfilterState processingChris Robinson2016-07-251-25/+17
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* Avoid using realloc in a number of placesChris Robinson2016-05-214-17/+19
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* Don't assume the "real" output buffer follows the dry bufferChris Robinson2016-05-171-15/+44
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* Improve reverb panning gains for "3D" output.Chris Robinson2016-05-171-8/+8
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* Get rid of an unnecessary copy of ALeffectPropsChris Robinson2016-05-1311-55/+54
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* Hold the effect and filter maps while handling effects and filtersChris Robinson2016-05-121-8/+0
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* Provide (mostly) lockless updates for effect slotsChris Robinson2016-05-1211-67/+83
| | | | | | | | | | | | | | | | | Similar to the listener, separate containers are provided atomically for the mixer thread to apply updates without needing to block, and a free-list is used to reuse container objects. A couple things to note. First, the lock is still used when the effect state's deviceUpdate method is called to prevent asynchronous calls to reset the device from interfering. This can be fixed by using the list lock in ALc.c instead. Secondly, old effect states aren't immediately deleted when the effect type changes (the actual type, not just its properties). This is because the mixer thread is intended to be real-time safe, and so can't be freeing anything. They are cleared away when updates reuse the container they were kept in, and they don't incur any extra processing cost, but there may be cases where the memory is kept around until the effect slot is deleted.
* Improve radius behavior with scaling of ambisonic coefficientsChris Robinson2016-04-245-15/+21
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* Don't look up the dry buffer's FrontCenter in the Dedicated effectChris Robinson2016-04-161-8/+4
| | | | | | The real FrontCenter output is used if it exists, and if it doesn't, it's unlikely the dry buffer will have it (and even if it does, it won't be any better than panning).
* More directly map coefficients for ambisonic mixing buffersChris Robinson2016-04-1510-44/+26
| | | | | | Instead of looping over all the coefficients for each channel with multiplies, when we know only one will have a non-0 factor for ambisonic mixing buffers, just index the one with a non-0 factor.
* Avoid mixing all coefficients together when only some are usedChris Robinson2016-04-155-21/+31
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* Use the 3D panned output for reverb with HQ decodingChris Robinson2016-03-311-3/+3
| | | | | | This is less than ideal, but matching each reverb line to a speaker with surround sound output is way too loud without the ambient volume scaling offered by the "direct" panning.
* Include any first-order scaling in the FOAOut coefficientsChris Robinson2016-03-253-16/+12
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* Mix Dedicated effects to the real output if possibleChris Robinson2016-03-251-6/+21
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* Add a specific output for first-order sourcesChris Robinson2016-03-223-3/+12
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* Add a workaround for a buggy modffChris Robinson2016-03-181-0/+15
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* Make sure enough reverb panning gains are definedChris Robinson2016-03-161-4/+4
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* Add a dual-band ambisonic decoderChris Robinson2016-03-151-8/+12
| | | | | | | | | | This uses a virtual B-Format buffer for mixing, and then uses a dual-band decoder for improved positional quality. This currently only works with first- order output since first-order input (from the AL_EXT_BFROMAT extension) would not sound correct when fed through a second- or third-order decoder. This also does not currently implement near-field compensation since near-field rendering effects are not implemented.
* Generalize GetChannelIdxByNameChris Robinson2016-03-101-2/+2
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* Allocate enough reverb panning gainsChris Robinson2016-03-101-2/+6
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* Organize the dry buffer properties into a structChris Robinson2016-03-0910-32/+40
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* Track the virtual and real output buffers ecplicitlyChris Robinson2016-03-091-15/+21
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