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path: root/OpenAL32/Include/alMain.h
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* Add queries to get the source offset with the device clockChris Robinson2017-12-031-0/+2
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* Avoid a separate function to query ambisonic mode supportChris Robinson2017-09-271-12/+8
| | | | | | Now FuMa and ACN channel orders are required, as are FuMa, SN3D, and N3D normalization schemes. An integer query (alcGetIntegerv) is added for the maximum ambisonic order.
* Re-update effect slots when context properties changeChris Robinson2017-09-271-1/+11
| | | | | Also keep all free property update structs together in the context instead of per-object.
* Don't update context and listener props unnecessarilyChris Robinson2017-09-271-0/+2
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* Update the context state properties separatelyChris Robinson2017-09-271-0/+5
| | | | | | | | | | | | | The context state properties are less likely to change compared to the listener state, and future changes may prefer more infrequent updates to the context state. Note that this puts the MetersPerUnit in as a context state, even though it's handled through the listener functions. Considering the infrequency that it's updated at (generally set just once for the context's lifetime), it makes more sense to put it there than with the more frequently updated listener properties. The aforementioned future changes would also prefer MetersPerUnit to not be updated unnecessarily.
* Manually save and restore the FPU rounding mode on WindowsChris Robinson2017-09-191-0/+3
| | | | | | | Apparently there is a bug with at least MinGW-W64 where fegetenv and fesetenv do not properly save and restore the FPU rounding mode, resulting in the rounding mode remaining as round-to-zero after certain function calls. I do not know if this also affects MSVC, but better safe than sorry for now.
* Add a front-stablizer config option for surround sound modesChris Robinson2017-07-311-0/+3
| | | | | | | | | | | | This improves a stereo (front-left + front-right) sound "image" by generating a front-center channel signal. Done correctly, it helps reduce the comb effects and phase errors associated with using only two speakers to simulate center sounds. Note that it shouldn't be used if the front-center channel is already included in the positional audio mix (the dialog effect is okay). In general, it may actually be better to exclude the front-center channel from the positional audio mix and use this to generate front-center output.
* Use macros to set and restore the mixer FPU modeChris Robinson2017-07-131-0/+11
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* Store the default effect slot in the contextChris Robinson2017-07-131-7/+4
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* Store the QSA backend's ExtraData in the wrapper structChris Robinson2017-06-291-2/+0
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* Remove the fastf2u conversion functionChris Robinson2017-06-271-5/+0
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* Clean up some messy rounding codeChris Robinson2017-06-261-0/+18
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* "Convert" the QSA backend to the new APIChris Robinson2017-06-181-19/+0
| | | | | | | | | | I say "convert" because it takes the lazy way and essentially just embeds the wrappers into the backend. It's done this way because I lack the means to check any changes, even syntactically. This also means the device's ExtraData field is still needed. However, this does mean all the backends are now using the new API. Code related to the old interface can now be removed.
* Make the dithering depth configurableChris Robinson2017-06-171-1/+1
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* Add a new compressor/limiterChris Robinson2017-05-271-2/+30
| | | | | | This is just for the output limiter right now, but in the future can be used for the compressor EFX effect. The parameters are also hardcoded, but can be made configurable after 1.18.
* Finalize ALC_SOFT_output_limiterChris Robinson2017-05-251-5/+0
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* Add an option to dither 8- and 16-bit outputChris Robinson2017-05-231-0/+4
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* Finalize AL_SOFT_source_spatializeChris Robinson2017-05-111-6/+0
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* Update AL_SOURCE_SPATIALIZE_SOFT valueChris Robinson2017-05-061-1/+1
| | | | | | Though it didn't strictly clash since it was for a different component (global state vs source property), 0x1213 was used by AL_RESAMPLER_NAME_SOFT. Probably best to avoid duplicate property values regardless.
* Calculate the output limiter gain using the RMSChris Robinson2017-05-051-1/+2
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* Start an extension to change the source's spatialize propertyChris Robinson2017-05-051-0/+6
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* Finalize AL_SOFT_source_resamplerChris Robinson2017-05-031-13/+0
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* Start an extension to toggle the output limiterChris Robinson2017-04-301-0/+5
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* Implement a limiter on the device outputChris Robinson2017-04-261-0/+2
| | | | | | This reduces the output volume when the mixed samples extend outside of -1,+1, to prevent excessive clipping. It can reduce the volume by -80dB in 50ms, and increase it by +80dB in 1s (it will not go below -80dB or above 0dB).
* Remove const from _Atomic vars to make Clang happyChris Robinson2017-04-211-0/+14
| | | | | | | | Clang does not allow using C11's atomic_load on const _Atomic variables. Previously it just disabled use of C11 atomics if atomic_load didn't work on a const _Atomic variable, but I think I'd prefer to have Clang use C11 atomics for the added features (more explicit memory ordering) even if it means a few instances of breaking const.
* Use more sensible values for the source resampler enumsChris Robinson2017-04-211-4/+4
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* Add a method to enumerate resamplersChris Robinson2017-04-211-0/+13
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* Allocate a new context's voices after updating the device paramsChris Robinson2017-04-191-0/+1
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* Use a different way to get the size of structs with flexible array membersChris Robinson2017-04-181-0/+6
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* Close some gaps in enum valuesChris Robinson2017-04-161-2/+2
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* Remove some unnecessary parenthesisChris Robinson2017-04-161-1/+1
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* Store the ambisonic order separate from the channel enumChris Robinson2017-04-121-8/+5
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* Convert the CoreAudio backend to the updated backend APIChris Robinson2017-04-091-3/+0
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* Reference count HRTFs and unload them when unusedChris Robinson2017-04-061-1/+1
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* Load HRTF files as neededChris Robinson2017-04-051-1/+1
| | | | | Currently only applies to external files, rather than embedded datasets. Also, HRTFs aren't unloaded after being loaded, until library shutdown.
* Store the loaded hrtf entry container in the enumerated hrtf entryChris Robinson2017-04-051-5/+6
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* Use an array of pointers for effects instead of a linked listChris Robinson2017-03-271-1/+1
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* Don't defer source state or offset changesChris Robinson2017-03-191-7/+1
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* Store the HRIR coeff pointer and delays directly in MixHrtfParamsChris Robinson2017-03-121-3/+4
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* Rework HRTF coefficient fadingChris Robinson2017-03-111-0/+1
| | | | | | | | | | | | | | | This improves fading between HRIRs as sources pan around. In particular, it improves the issue with individual coefficients having various rounding errors in the stepping values, as well as issues with interpolating delay values. It does this by doing two mixing passes for each source. First using the last coefficients that fade to silence, and then again using the new coefficients that fade from silence. When added together, it creates a linear fade from one to the other. Additionally, the gain is applied separately so the individual coefficients don't step with rounding errors. Although this does increase CPU cost since it's doing two mixes per source, each mix is a bit cheaper now since the stepping is simplified to a single gain value, and the overall quality is improved.
* Allocate as many channels for DirectHrtfState as neededChris Robinson2017-03-111-2/+4
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* Dynamically allocate the device's HRTF stateChris Robinson2017-03-101-13/+14
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* Implement NFC filters for Ambisonic renderingChris Robinson2017-03-101-0/+7
| | | | | | | | | | | | | | NFC filters currently only work when rendering to ambisonic buffers, which includes HQ rendering and ambisonic output. There are two new config options: 'decoder/nfc' (default on) enables or disables use of NFC filters globally, and 'decoder/nfc-ref-delay' (default 0) specifies the reference delay parameter for NFC-HOA rendering with ambisonic output (a value of 0 disables NFC). Currently, NFC filters rely on having an appropriate value set for AL_METERS_PER_UNIT to get the correct scaling. HQ rendering uses the averaged speaker distances as a control/reference, and currently doesn't correct for individual speaker distances (if the speakers are all equidistant, this is fine, otherwise per-speaker correction should be done as well).
* Dynamically allocate the channel delay buffersChris Robinson2017-02-281-2/+2
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* Remove unused function declarationsChris Robinson2017-02-281-3/+0
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* Start a ALC_SOFT_loopback2 extensionChris Robinson2017-02-281-5/+28
| | | | | | | | | | | | | | | | | | | | | | This extends the base ALC_SOFT_loopback extension with support for B-Format. When ALC_FORMAT_CHANNELS_SOFT is set to ALC_BFORMAT3D_SOFT, then additional attributes must be specified. ALC_AMBISONIC_LAYOUT_SOFT must be set to ALC_ACN_SOFT or ALC_FUMA_SOFT for the desired channel layout, ALC_AMBISONIC_SCALING_SOFT must be set to ALC_N3D_SOFT, ALC_SN3D_SOFT, or ALC_FUMA_SOFT for the desired channel scaling/normalization scheme, and ALC_AMBISONIC_ORDER_SOFT must be set to an integer value greater than 0 for the ambisonic order (maximum allowed is implementation-dependent). Note that the number of channels required for ALC_BFORMAT3D_SOFT is dependent on the ambisonic order. The number of channels can be calculated by: num_channels = (order+1) * (order+1); /* or pow(order+1, 2); */ In addition, a new alcIsAmbisonicFormatSupportedSOFT function allows apps to determine which layout/scaling/order combinations are supported by the loopback device. For example, alcIsAmbisonicFormatSupported(device, ALC_ACN_SOFT, ALC_SN3D_SOFT, 2) will check if 2nd order AmbiX (ACN layout and SN3D scaling) rendering is supported for ALC_BFORMAT3D_SOFT output.
* Use separate enums for the ambisonic channel order and normalizationChris Robinson2017-02-271-6/+14
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* Dynamically allocate the ALsource Send[] arrayChris Robinson2017-02-211-1/+1
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* Make the voices' Send[] array dynamically sizedChris Robinson2017-02-211-1/+1
| | | | | 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 Robinson2017-02-191-0/+12
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