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path: root/OpenAL32/Include/alMain.h
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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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* Make ALsourceProps' Send array dynamically sizedChris Robinson2017-02-141-0/+9
| | | | | | 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.
* Remove a couple context lock wrapper functionsChris Robinson2017-02-071-6/+0
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* Remove __android_log_print calls for TRACEREFChris Robinson2017-01-271-1/+0
| | | | | TRACEREFs aren't normally important, and for as often as it happens, the added function calls are wasteful even if they end up doing nothing.
* Also log to __android_log_print on AndroidChris Robinson2017-01-261-0/+11
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* Move the B-Format HRTF virtual speaker stuff to InitHrtfPanningChris Robinson2017-01-181-12/+20
| | | | | This keeps the decoder matrices and coefficient mapping together for if it changes in the future.
* Use ALsizei in more placesChris Robinson2017-01-181-8/+8
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* Use ALsizei for sizes and offsets with the mixerChris Robinson2017-01-161-5/+5
| | | | | | 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 Robinson2017-01-151-2/+2
| | | | | | 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).
* Convert the SndIO backend to the updated APIChris Robinson2016-12-211-3/+0
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* Make some pointer-to-array parameters constChris Robinson2016-10-041-0/+42
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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 Robinson2016-09-241-0/+2
| | | | | | | | | 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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* 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.
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