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* Make ReleaseContext return if any contexts still remainChris Robinson2017-03-281-10/+25
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* Use an array of pointers for effects instead of a linked listChris Robinson2017-03-271-1/+11
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* Fix setting Ambi formats for loopback devicesChris Robinson2017-03-211-1/+1
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* Use an atomic flag to test if a source needs to updateChris Robinson2017-03-201-1/+1
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* Don't defer source state or offset changesChris Robinson2017-03-191-32/+3
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* Fix alcGetInteger64vSOFT to handle ambisonic attributesChris Robinson2017-03-181-14/+28
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* Replace a couple ALuint with ALsizeiChris Robinson2017-03-171-2/+2
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* Remove a couple unneeded typedefsChris Robinson2017-03-141-23/+25
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* Dynamically allocate the device's HRTF stateChris Robinson2017-03-101-37/+42
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* Fix building without C11Chris Robinson2017-03-101-1/+2
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* Implement NFC filters for Ambisonic renderingChris Robinson2017-03-101-0/+20
| | | | | | | | | | | | | | 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).
* Remove an unnecessary variableChris Robinson2017-03-091-2/+1
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* Make the voice's source pointer atomicChris Robinson2017-03-051-1/+1
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* Increment MixCount in UpdateClockBaseChris Robinson2017-02-281-1/+4
| | | | This is to protect clocktime reads since the backend lock won't protect it.
* Dynamically allocate the channel delay buffersChris Robinson2017-02-281-0/+42
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* Start a ALC_SOFT_loopback2 extensionChris Robinson2017-02-281-50/+190
| | | | | | | | | | | | | | | | | | | | | | 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 a variable counter for an array size limitChris Robinson2017-02-281-21/+13
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* Print WARNs when a device or context error is generatedChris Robinson2017-02-271-0/+1
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* Use separate enums for the ambisonic channel order and normalizationChris Robinson2017-02-271-6/+18
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* Move the current buffer queue entry and play position to the voiceChris Robinson2017-02-271-12/+14
| | | | | | | | | | | | | | 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.
* Reduce the default period count to 3Chris Robinson2017-02-221-1/+1
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* Make the "sends" config option act as a limitChris Robinson2017-02-211-8/+16
| | | | | Instead of forcing the device to always use the specified send count, it simply limits requests to it.
* Increase the default effect slot and send countChris Robinson2017-02-211-13/+15
| | | | | | | | | | The default number of auxiliary effect slots is now 64. This can still be raised by the config file without a hard maximum, but incurs processing cost for each effect slot generated by the app. The default number of source sends is now actually 2, as per the EFX docs. However, it can be raised up to 16 via ALC_MAX_AUXILIARY_SENDS attribute requests, rather than the previous 4.
* Dynamically allocate the ALsource Send[] arrayChris Robinson2017-02-211-44/+58
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* Interleave the voice and source property objectsChris Robinson2017-02-211-13/+12
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* Make the voices' Send[] array dynamically sizedChris Robinson2017-02-211-12/+28
| | | | | 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.
* Avoid duplicating device buffer layout logicChris Robinson2017-02-201-36/+13
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* Don't use periphonic FOA when the HOA decoder is not periphonicChris Robinson2017-02-191-21/+29
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* Always lock the device backend before calling aluMixDataChris Robinson2017-02-181-0/+4
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* Make ALsourceProps' Send array dynamically sizedChris Robinson2017-02-141-9/+90
| | | | | | 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 Robinson2017-02-131-2/+1
| | | | | Since it's modified by the mixer when playback is ended, a plain struct member isn't safe.
* Remove a couple context lock wrapper functionsChris Robinson2017-02-071-3/+0
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* Convert the OpenSL backend to the new backend APIChris Robinson2017-02-071-1/+1
| | | | | This also removes the buffer queue callback's call to aluMixData, which could potentially block on a mutex.
* Fix for NULL JNIEnvChris Robinson2017-02-051-0/+6
| | | | Which can happen with native-only apps
* Get the JavaVM handle on Android targetsChris Robinson2017-01-261-0/+64
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* Use ALsizei in more placesChris Robinson2017-01-181-4/+4
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* Use second-order ambisonics for basic HRTF renderingChris Robinson2017-01-151-3/+8
| | | | | | 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).
* Add missing AL_EFFECTSLOT_ properties for al(c)GetEnumValueChris Robinson2017-01-051-0/+5
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* Convert the SndIO backend to the updated APIChris Robinson2016-12-211-1/+1
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* Trace the commit ID and branch the library was built fromChris Robinson2016-12-211-0/+4
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* Use separate macros for atomics that don't take a memory orderChris Robinson2016-12-201-34/+36
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* Only send source updates for sources that have updatedChris Robinson2016-11-231-1/+2
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* Remove the non-atomic COMPARE_EXCHANGE macroChris Robinson2016-11-221-33/+46
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* Stop using almemory_order_consumeChris Robinson2016-11-171-2/+2
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* Finalize AL_SOFT_gain_clamp_exChris Robinson2016-10-031-1/+1
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* Add a volume-adjust config option to adjust the source output volumeChris Robinson2016-09-241-0/+17
| | | | | | | | | 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 use of DECL_CONSTChris Robinson2016-09-061-4/+4
| | | | | No idea if it was really gaining us anything, but removing it fixes a crash I was getting with libs built with Clang.
* 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 Robinson2016-08-281-0/+1
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* Allow sources to play while alcSuspendContext is in effectChris Robinson2016-08-261-3/+3
| | | | | | | | | | | | | | 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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