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* Add some more 'restrict' keywordsChris Robinson2016-10-061-1/+1
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* Remove an unused structChris Robinson2016-10-051-6/+0
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* Pass current and target gains directly for mixingChris Robinson2016-10-051-2/+3
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* Make some pointer-to-array parameters constChris Robinson2016-10-042-2/+45
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* Finalize AL_SOFT_gain_clamp_exChris Robinson2016-10-031-5/+0
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* Add a volume-adjust config option to adjust the source output volumeChris Robinson2016-09-242-5/+7
| | | | | | | | | 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.
* Remove some more unnecessary volatilesChris Robinson2016-09-241-5/+5
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* Call ALfilterState_processC directlyChris Robinson2016-09-122-5/+2
| | | | | | 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.
* Mark a global variable declaration as externChris Robinson2016-09-111-1/+1
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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 Robinson2016-09-063-14/+12
| | | | | 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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* Remove unnecessary constsChris Robinson2016-08-311-4/+4
| | | | They were causing GCC's built-in atomic cmpxchg to complain.
* Reorganize sample type conversion functionsChris Robinson2016-08-311-286/+129
| | | | To help avoid redundant manual definitions.
* Initialize some enums to dummy valuesChris Robinson2016-08-311-8/+9
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* Remove the upper limit from AL_MIN_GAIN and AL_MAX_GAINChris Robinson2016-08-291-2/+2
| | | | As per the current AL_SOFT_gain_clamp_ex proposal.
* Add a query for the maximum source gain limitChris Robinson2016-08-282-0/+31
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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 Robinson2016-08-263-8/+19
| | | | | | | | | | | | | | 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.
* Avoid directly replacing the effect slot Update pointerChris Robinson2016-08-251-14/+10
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* Properly defer effect slot changesChris Robinson2016-08-252-11/+31
| | | | | | | | 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.
* Track all references for effect statesChris Robinson2016-08-251-21/+47
| | | | | | | | This allows us to not have to play around with trying to avoid duplicate state pointers, since the reference count will ensure they're deleted as appropriate. The only caveat is that the mixer is not allowed to decrement references, since that can cause the object to be freed (which the mixer code is not allowed to do).
* Add a ref count to ALeffectStateChris Robinson2016-08-253-0/+16
| | | | | 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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* Make a function definition staticChris Robinson2016-08-231-1/+1
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* Hold updates for both listener and source updatesChris Robinson2016-08-231-11/+0
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* Don't pass the context's distance model as the source'sChris Robinson2016-08-234-13/+18
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* Avoid resupplying unneeded source updatesChris Robinson2016-08-234-62/+38
| | | | | 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.
* Avoid checking DeferUpdates for each source state changeChris Robinson2016-08-081-8/+24
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* Use al_calloc/al_free in more placesChris Robinson2016-08-032-9/+9
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* Add 'restrict' to another parameterChris Robinson2016-08-032-2/+2
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* Don't store the looping state in the voiceChris Robinson2016-07-312-13/+22
| | | | | 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.
* Avoid a NULL deref when creating 0 auxiliary effect slotsChris Robinson2016-07-261-12/+9
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* Remove the last use of ALfilterState_processSingleChris Robinson2016-07-262-18/+0
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* Remove broken autowah effect codeChris Robinson2016-07-265-21/+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 Robinson2016-07-262-4/+3
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* Increase the default thunk array sizeChris Robinson2016-07-251-1/+1
| | | | | This value should be enough to hold IDs for most apps without needing to reallocate it, while not being unnecessarily large (4KB).
* 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 Robinson2016-07-133-18/+16
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* Store the voice output buffers separate from the paramsChris Robinson2016-07-112-6/+10
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