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* Use unsigned int shifts for device flagsChris Robinson2016-09-071-5/+5
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* Make the SelectMixer function sharableChris Robinson2016-09-061-0/+2
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* Remove use of DECL_CONSTChris Robinson2016-09-062-10/+8
| | | | | No idea if it was really gaining us anything, but removing it fixes a crash I was getting with libs built with Clang.
* Use a predefined identity matrixChris Robinson2016-09-051-0/+1
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* Rename MatrixMixerFunc to RowMixerFuncChris Robinson2016-09-021-3/+3
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* Add a query for the maximum source gain limitChris Robinson2016-08-281-0/+5
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* Clamp the maximum mixing gain boost to 16Chris Robinson2016-08-271-0/+2
| | | | | | The combined source and listener gains now can't exceed a multiplier of 16 (~24dB). This is to avoid mixes getting out of control with large volume boosts, which reduces the effective precision given by floating-point.
* Allow sources to play while alcSuspendContext is in effectChris Robinson2016-08-261-1/+7
| | | | | | | | | | | | | | This appears to be how Creative's Windows drivers handle it, and is necessary for at least the Windows version of UT2k4 (otherwise it tries to play a source while suspended, checks and sees it's stopped, then kills it before it's given a chance to start playing). Consequently, the internal properties it gets mixed with are determined by what the source properties are at the time of the play call, and the listener properties at the time of the suspend call. This does not change alDeferUpdatesSOFT, which will still hold the play state change until alProcessUpdatesSOFT.
* Properly defer effect slot changesChris Robinson2016-08-251-0/+3
| | | | | | | | Note that this now also causes all playing sources to update when an effect slot is updated. This is a bit wasteful, as it should only need to re-update sources that are using the effect slot (and only when a relevant property is changed), but it's good enough. Especially with deferring since all playing sources are going to get updated on the process call anyway.
* Add a ref count to ALeffectStateChris Robinson2016-08-252-0/+8
| | | | | This is mostly just reorganizing the effects to call the Construct method which initializes the ref count.
* Combine related members into a structChris Robinson2016-08-241-10/+12
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* Don't pass the context's distance model as the source'sChris Robinson2016-08-231-0/+5
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* Avoid resupplying unneeded source updatesChris Robinson2016-08-231-34/+37
| | | | | The source's voice holds a copy of the last properties it received, so listener updates can make sources recalculate internal properties from that stored copy.
* Use a more specialized mixer function for B-Format to HRTFChris Robinson2016-08-122-2/+6
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* Decode directly from B-Format to HRTF instead of a cubeChris Robinson2016-08-111-2/+3
| | | | | | | | | | | | | | | | | | | | | Last time this attempted to average the HRIRs according to their contribution to a given B-Format channel as if they were loudspeakers, as well as averaging the HRIR delays. The latter part resulted in the loss of the ITD (inter-aural time delay), a key component of HRTF. This time, the HRIRs are averaged similar to above, except instead of averaging the delays, they're applied to the resulting coefficients (for example, a delay of 8 would apply the HRIR starting at the 8th sample of the target HRIR). This does roughly double the IR length, as the largest delay is about 35 samples while the filter is normally 32 samples. However, this is still smaller the original data set IR (which was 256 samples), it also only needs to be applied to 4 channels for first-order ambisonics, rather than the 8-channel cube. So it's doing twice as much work per sample, but only working on half the number of samples. Additionally, since the resulting HRIRs no longer rely on an extra delay line, a more efficient HRTF mixing function can be made that doesn't use one. Such a function can also avoid the per-sample stepping parameters the original uses.
* Add 'restrict' to another parameterChris Robinson2016-08-031-1/+1
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* Don't store the looping state in the voiceChris Robinson2016-07-311-4/+2
| | | | | Certain operations on the buffer queue depend on the loop state to behave properly, so it should not be deferred until the async voice update occurs.
* Remove DevFmtBFormat3D, which is covered by DevFmtAmbi1Chris Robinson2016-07-311-1/+1
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* Add an option to specify the ambisonic output configurationChris Robinson2016-07-311-0/+12
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* Add a stand-alone upsampler for higher-order ambisonic oputputChris Robinson2016-07-301-1/+4
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* Add a config to output first-, second-, or third-order ambisonicsChris Robinson2016-07-291-0/+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 Robinson2016-07-261-17/+0
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* Remove broken autowah effect codeChris Robinson2016-07-262-10/+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.
* Add some more restrict keywordsChris Robinson2016-07-261-2/+2
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* Rename input_gain to b0Chris Robinson2016-07-261-3/+2
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* Make a MAX_AMBI2D_COEFFS macro instead of a magic numberChris Robinson2016-07-171-1/+8
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* Modify bs2b_cross_feed to do multiple samples at onceChris Robinson2016-07-131-36/+5
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* Move the input channel array out of the DirectParams and SendParamsChris Robinson2016-07-132-15/+13
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* Store the voice output buffers separate from the paramsChris Robinson2016-07-112-6/+10
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* Reorder some source fieldsChris Robinson2016-07-071-5/+5
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* Avoid using memcpy to copy a single structChris Robinson2016-07-061-14/+12
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* Remove a slightly outdated commentChris Robinson2016-07-061-1/+0
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* Remove the VirtOut buffer aliasChris Robinson2016-07-051-6/+3
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* Rename MaxNoOfSources for consistencyChris Robinson2016-06-081-1/+1
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* Only define 8 HRTF filter states and params for the deviceChris Robinson2016-06-041-2/+2
| | | | | It will only be used with a cube channel setup, so there's no need to have one for every possible output channel.
* Add property queries to get the device latency with the clockChris Robinson2016-06-031-0/+2
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* Remove some unnecessary volatile keywordsChris Robinson2016-06-032-33/+33
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* Make a function staticChris Robinson2016-06-011-1/+0
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* Use a macro to specify the ambisonic periphonic channel maskChris Robinson2016-06-011-1/+10
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* Use SSE for applying the HQ B-Format decoder matricesChris Robinson2016-05-311-0/+3
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* Hold the effectslot map lock while handling itChris Robinson2016-05-291-2/+11
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* Use a linked list for active effect slotsChris Robinson2016-05-292-1/+3
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* Use a specific lock for the backend's stop/reset/play callsChris Robinson2016-05-271-0/+2
| | | | | This helps protect against the device changing unexpectedly from multiple threads, instead of using the global list/library lock.
* Avoid using realloc in a number of placesChris Robinson2016-05-211-0/+1
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* Make the source position calues atomicChris Robinson2016-05-191-2/+2
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* Avoid redundantly storing distance model settingsChris Robinson2016-05-172-7/+1
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* Use floats for the listener transformsChris Robinson2016-05-162-26/+1
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* Don't store the source's update method with the voiceChris Robinson2016-05-162-6/+0
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* Avoid separate updates to sources that should apply togetherChris Robinson2016-05-152-2/+6
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* Avoid using a flag to specify if the effect state needs to be updatedChris Robinson2016-05-151-3/+1
| | | | | This fixes a potential missed state change if an update with a new state got replaced with one that doesn't.
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