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path: root/OpenAL32/Include/alMain.h
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* Use unsigned int shifts for device flagsChris Robinson2016-09-071-5/+5
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* Remove use of DECL_CONSTChris Robinson2016-09-061-6/+4
| | | | | No idea if it was really gaining us anything, but removing it fixes a crash I was getting with libs built with Clang.
* Add a query for the maximum source gain limitChris Robinson2016-08-281-0/+5
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* 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.
* Add a ref count to ALeffectStateChris Robinson2016-08-251-0/+6
| | | | | 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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* Use a more specialized mixer function for B-Format to HRTFChris Robinson2016-08-121-2/+2
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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.
* 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.
* Make a MAX_AMBI2D_COEFFS macro instead of a magic numberChris Robinson2016-07-171-1/+8
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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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* Use a macro to specify the ambisonic periphonic channel maskChris Robinson2016-06-011-1/+10
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* Use a linked list for active effect slotsChris Robinson2016-05-291-1/+1
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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 separate updates to sources that should apply togetherChris Robinson2016-05-151-0/+6
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* Provide asynchronous property updates for sourcesChris Robinson2016-05-141-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | This necessitates a change in how source updates are handled. Rather than just being able to update sources when a dependent object state is changed (e.g. a listener gain change), now all source updates must be proactively provided. Consequently, apps that do not utilize any deferring (AL_SOFT_defer_updates or alcSuspendContext/alcProcessContext) may utilize more CPU since it'll be filling out more update containers for the mixer thread to use. The upside is that there's less blocking between the app's calling thread and the mixer thread, particularly for vectors and other multi-value properties (filters and sends). Deferring behavior when used is also improved, since updates that shouldn't be applied yet are simply not provided. And when they are provided, the mixer doesn't have to ignore them, meaning the actual deferring of a context doesn't have to synchrnously force an update -- the process call will send any pending updates, which the mixer will apply even if another deferral occurs before the mixer runs, because it'll still be there waiting on the next mixer invocation. There is one slight bug introduced by this commit. When a listener change is made, or changes to multiple sources while updates are being deferred, it is possible for the mixer to run while the sources are prepping their updates, causing some of the source updates to be seen before the other. This will be fixed in short order.
* Use a lockless method for updating listener and context propertiesChris Robinson2016-05-111-4/+3
| | | | | | | | | | | This uses a separate container to provide the relevant properties to the internal update method, using atomic pointer swaps. A free-list is used to avoid having too many individual containers. This allows the mixer to update the internal listener properties without requiring the lock to protect against async updates. It also allows concurrent read access to the user-facing property values, even the multi-value ones (e.g. the vectors).
* Add a comment about CoeffCount being 0Chris Robinson2016-05-101-1/+3
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* Start AL_SOFT_buffer_samples2 as a replacement for AL_SOFT_buffer_samplesChris Robinson2016-04-251-0/+73
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* Increase max output channels to 16Chris Robinson2016-04-191-1/+8
| | | | This also enables fully periphonic 3rd order HQ decoding.
* Avoid storing channel names for the dry bufferChris Robinson2016-04-161-2/+0
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* More directly map coefficients for ambisonic mixing buffersChris Robinson2016-04-151-4/+12
| | | | | | 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 Robinson2016-04-151-0/+4
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* Avoid unnecessary loops for setting up effect slot b-format buffer mixingChris Robinson2016-04-141-0/+4
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* Split aluInitPanning into separate functions for HRTF or UHJChris Robinson2016-04-141-3/+5
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* Remove the old ringbuffer implementationChris Robinson2016-03-301-7/+0
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* Move the aligned malloc functions to the common libChris Robinson2016-03-291-5/+1
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* Include any first-order scaling in the FOAOut coefficientsChris Robinson2016-03-251-2/+0
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* Allow second-order HQ decodingChris Robinson2016-03-231-1/+6
| | | | | | | | Could really do with some optimizations to the mixing gain calculations. For ambisonic targets, the coefficients will only have 1 non-0 entry for each output, so the double loop in unnecessarily wasteful. Similarly, most uses won't need a full height encoding either, so a horizontal-only or mixed-order target could reduce the number of channels.
* Add a specific output for first-order sourcesChris Robinson2016-03-221-0/+9
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* Rename the BFormat channels to AuxChris Robinson2016-03-161-4/+4
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* Add a dual-band ambisonic decoderChris Robinson2016-03-151-0/+3
| | | | | | | | | | 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.
* Add a loader for ambdec filesChris Robinson2016-03-141-1/+2
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* Generalize GetChannelIdxByNameChris Robinson2016-03-101-6/+6
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* Keep track of the real output's channel namesChris Robinson2016-03-101-0/+2
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* Organize the dry buffer properties into a structChris Robinson2016-03-091-19/+25
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* Track the virtual and real output buffers ecplicitlyChris Robinson2016-03-091-1/+12
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* Add an option for pair-wise stereo panningChris Robinson2016-02-261-5/+7
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* Use 2-channel UHJ for stereo outputChris Robinson2016-02-261-0/+3
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* Remove an unused functionChris Robinson2016-02-231-2/+0
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* Use an 8-channel cube for HRTF's virtual format.Chris Robinson2016-02-201-0/+9
| | | | | | There were phase issues caused by applying HRTF directly to the B-Format channels, since the HRIR delays were all averaged which removed the inter-aural time-delay, which in turn removed significant spatial information.
* Calculate HRTF stepping params right before mixingChris Robinson2016-02-141-2/+0
| | | | | This means we track the current params and the target params, rather than the target params and the stepping. This closer matches the non-HRTF mixers.
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