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* Add a boolean to specify if a voice should be playingChris Robinson2017-03-021-10/+9
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* Dynamically allocate the channel delay buffersChris Robinson2017-02-281-6/+6
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* Move the current buffer queue entry and play position to the voiceChris Robinson2017-02-271-6/+3
| | | | | | | | | | | | | | 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.
* Ensure a non-playing or -paused source does not use a mixing voiceChris Robinson2017-02-251-1/+4
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* Alter how panpot/pair-wise panning worksChris Robinson2017-02-231-28/+10
| | | | | | | | | | | | | | | | | | | 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.
* Rename stereo-panning option to stereo-encodingChris Robinson2017-02-221-4/+4
| | | | Also rename the 'paired' value to 'panpot', and make it the default.
* Limit filter gains to -24dBChris Robinson2017-02-221-32/+26
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* Reduce some codeChris Robinson2017-02-211-61/+33
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* Make the voices' Send[] array dynamically sizedChris Robinson2017-02-211-11/+11
| | | | | 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.
* Apply distance compensation when writing to the outputChris Robinson2017-02-191-13/+38
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* Don't use periphonic FOA when the HOA decoder is not periphonicChris Robinson2017-02-191-4/+3
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* Always lock the device backend before calling aluMixDataChris Robinson2017-02-181-2/+0
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* Reorganize ALvoice membersChris Robinson2017-02-151-81/+87
| | | | | This places the Send[] array at the end of the struct, making it easier to handle dynamically.
* Make ALsourceProps' Send array dynamically sizedChris Robinson2017-02-141-11/+11
| | | | | | 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.
* Make the source state atomicChris Robinson2017-02-131-4/+6
| | | | | Since it's modified by the mixer when playback is ended, a plain struct member isn't safe.
* Put BsincState in a generic unionChris Robinson2017-02-131-2/+2
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* Fix more uses of unsigned sizes and offsetsChris Robinson2017-02-101-5/+5
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* Replace some ALvoid with voidChris Robinson2017-01-181-2/+2
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* Use ALsizei in more placesChris Robinson2017-01-181-1/+1
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* Pass the left and right buffers to the hrtf mixers directlyChris Robinson2017-01-171-14/+16
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* Use ALsizei for sizes and offsets with the mixerChris Robinson2017-01-161-1/+1
| | | | | | Unsigned 32-bit offsets actually have some potential overhead on 64-bit targets for pointer/array accesses due to rules on integer wrapping. No idea how much impact it has in practice, but it's nice to be correct about it.
* Use second-order ambisonics for basic HRTF renderingChris Robinson2017-01-151-0/+7
| | | | | | This should improve positional quality for relatively low cost. Full HRTF rendering still only uses first-order since the only use of the dry buffer there is for first-order content (B-Format buffers, effects).
* Avoid duplicating code using a macroChris Robinson2016-12-211-38/+3
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* Use separate macros for atomics that don't take a memory orderChris Robinson2016-12-201-7/+7
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* Update some atomic memory orderingChris Robinson2016-11-211-6/+9
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* Don't interpolate between nearest HRIRsChris Robinson2016-10-091-4/+4
| | | | | | | | It still fades between HRIRs when it changes, but now it selects the nearest one instead of blending the nearest four. Due to the minimum-phase nature of the HRIRs, interpolating between delays lead to some oddities which are exasperated by the fading (and the fading is needed to avoid clicks and pops, and smooth out changes).
* Make some pointer-to-array parameters constChris Robinson2016-10-041-7/+6
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* Add a volume-adjust config option to adjust the source output volumeChris Robinson2016-09-241-1/+1
| | | | | | | | | 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.
* Use a predefined identity matrixChris Robinson2016-09-051-0/+7
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* Correct a comment about B-Format conversionChris Robinson2016-09-051-1/+1
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* Clamp the maximum mixing gain boost to 16Chris Robinson2016-08-271-10/+11
| | | | | | 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.
* Properly defer effect slot changesChris Robinson2016-08-251-3/+5
| | | | | | | | 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 Robinson2016-08-251-9/+6
| | | | | | | | 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).
* Consolidate duplicate codeChris Robinson2016-08-241-39/+23
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* Combine related members into a structChris Robinson2016-08-241-7/+7
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* Don't pass the context's distance model as the source'sChris Robinson2016-08-231-1/+6
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* Avoid resupplying unneeded source updatesChris Robinson2016-08-231-28/+51
| | | | | 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 Robinson2016-08-121-11/+8
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* Decode directly from B-Format to HRTF instead of a cubeChris Robinson2016-08-111-1/+1
| | | | | | | | | | | | | | | | | | | | | 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.
* Constify some variablesChris Robinson2016-08-041-2/+2
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* Don't store the looping state in the voiceChris Robinson2016-07-311-2/+0
| | | | | 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.
* Add a stand-alone upsampler for higher-order ambisonic oputputChris Robinson2016-07-301-0/+7
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* Modify bs2b_cross_feed to do multiple samples at onceChris Robinson2016-07-131-22/+17
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* Move the input channel array out of the DirectParams and SendParamsChris Robinson2016-07-131-55/+56
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* Store the voice output buffers separate from the paramsChris Robinson2016-07-111-20/+20
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* Avoid function calls to get the HRTF sample rate and IR sizeChris Robinson2016-07-071-1/+1
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* Remove the VirtOut buffer aliasChris Robinson2016-07-051-10/+11
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* Ensure voices has been updated once before mixing themChris Robinson2016-06-161-1/+2
| | | | | | | | Sometimes the mixer is temporarily prevented from applying updates, when multiple sources need to be updated simultaneously for example, but does not prevent mixing. If the mixer runs during that time and a voice was just started, it would've mixed the voice without any internal properties being set for it.
* Use a linked list for active effect slotsChris Robinson2016-05-291-16/+24
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* Increment the device's mix count closer to the mixing loopsChris Robinson2016-05-231-3/+2
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