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* Use ComputeAngleGains for multi-channel sources and remove the unused lookup ↵Chris Robinson2012-04-292-5/+0
| | | | table stuff
* Remove the stereodup optionChris Robinson2012-04-291-2/+0
| | | | There's better ways to do it now with "wide" sound sources.
* Add a method to calculate gains given a sound point and its half-width, and ↵Chris Robinson2012-04-281-0/+2
| | | | | | | | | use it for reverb The half-width ranges from 0 to pi, and essentially specifies the coverage area around the listener. At 0, it's an infinitely small point sound and behaves like a usual panning sound. At pi/2 it covers half the area, and at pi it covers the whole area.
* Flip the parameters to aluCart2LUTpos, so it behaves a bit more like atan2Chris Robinson2012-04-281-1/+1
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* Pass the direct/send params to the mixer instead of accessing through the sourceChris Robinson2012-04-281-0/+4
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* Move the HrtfParams into the DirectParamsChris Robinson2012-04-281-3/+3
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* Move the direct and send params into separate different typesChris Robinson2012-04-281-23/+33
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* Use separate methods for the dry and wet mixing loopsChris Robinson2012-04-272-7/+13
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* Store the speaker angles in the deviceChris Robinson2012-04-271-0/+1
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* Reorganize some ALsource fieldsChris Robinson2012-04-271-20/+30
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* The mixer always uses floatsChris Robinson2012-04-261-1/+1
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* Use a separate loop to actually update DataPosFrac and DataPosIntChris Robinson2012-04-261-2/+1
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* Add and fixup some ALsource commentsChris Robinson2012-04-261-25/+42
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* Add try/catch-like macros to handle errors, and convert alSource.c to use themChris Robinson2012-04-231-0/+47
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* Remove hungarian notation from the device and context structsChris Robinson2012-04-191-3/+3
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* Remove an unused fieldChris Robinson2012-04-191-2/+0
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* Use a consistent name for the self-id fieldChris Robinson2012-04-194-8/+8
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* Fix comment and rename the buffer's self-idChris Robinson2012-04-191-3/+4
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* Remove hungarian notation from sourcesChris Robinson2012-04-191-16/+16
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* Use a double to store the source offset and don't use milliseconds for secondsChris Robinson2012-04-161-2/+2
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* Finalize ALC_SOFT_loopbackChris Robinson2012-03-151-32/+0
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* Rename GetReverbEffect to LoadReverbPresetChris Robinson2012-03-131-1/+1
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* Use enums to specify the default channel configuration and sample typeChris Robinson2012-03-131-2/+6
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* Pass the device to InitializeEffect and return the error enum from itChris Robinson2012-03-131-1/+1
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* Pass a device to the effect update functionsChris Robinson2012-03-131-1/+1
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* Finalize AL_SOFT_direct_channelsChris Robinson2012-03-121-5/+0
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* Use a separate backend callback to start playback of the deviceChris Robinson2012-03-051-0/+2
| | | | | | | | | | | | This allows us to properly update the ALCdevice and its resources with the new parameters before starting playback, instead of expecting the mixer to block and wait after it has begun. This also lets us avoid holding the device lock while resetting and starting the device, which helps prevent lock inversion on some backends (ie, one thread locking A then B, and another thread locking B then A), ultimately allowing certain backends to asynchronously update the ALCdevice without risk of lockup. Capture still has issues here, however.
* Make alcLoopbackOpenDeviceSOFT take a standard "device" name for a parameterChris Robinson2012-03-011-2/+2
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* Don't use GUIDs to ID mmdevapi devices, and don't enumerate if not neededChris Robinson2012-03-011-0/+2
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* Use an enum for the device typeChris Robinson2012-02-231-3/+8
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* Always use "OpenAL Soft" for the short device enumeration listChris Robinson2012-02-201-2/+0
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* Use memset to clear the bs2b filter's last sample and remove the unused ↵Chris Robinson2012-02-191-3/+0
| | | | is_clear function
* Add a COUNTOF macro to get the number of entries in a static arrayChris Robinson2012-02-191-0/+1
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* Move the endian test macro to alMain.hChris Robinson2012-02-151-0/+7
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* Add a device parameter to alcLoopbackOpenDeviceSOFTChris Robinson2012-02-151-2/+2
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* Add a device request flag for the sample typeChris Robinson2012-02-151-0/+2
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* Remove parameter names from typedefsChris Robinson2012-02-151-2/+2
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* Define a minimum output sample rate and warn if the config option goes below itChris Robinson2012-02-151-0/+1
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* Support signed and unsigned 32-bit int outputChris Robinson2012-02-141-0/+2
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* Use more proper enum names for the resamplerChris Robinson2012-02-122-8/+6
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* Use a proper enum value for setting the default distance modelChris Robinson2012-02-121-1/+3
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* Rename non_virtual_channels to direct_channelsChris Robinson2012-02-092-4/+4
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* Check for a 32-bit x86 processor before using MSVC's asm intrinsicsChris Robinson2012-02-031-1/+1
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* Only use fast float-to-int GCC assembly for x86 and x86_64 targetsChris Robinson2012-02-031-1/+1
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* Avoid duplicating some initialization codeChris Robinson2012-01-202-3/+5
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* Add the ability to specify a preset for forced reverbChris Robinson2012-01-191-1/+1
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* Add a global option to apply a reverb effect on source send 0Chris Robinson2012-01-193-0/+5
| | | | | | A special slot on the device is created and processed, so it can be shared across all contexts on the device. Sources that don't have a slot set on send 0 will use this special slot instead.
* Finalize AL_SOFT_buffer_samplesChris Robinson2012-01-101-70/+0
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* Remove the unnecessary AL_MULAW_SOFT, AL_ALAW_SOFT, and AL_IMA4_SOFT definesChris Robinson2012-01-102-6/+3
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* Append _SOFT to the AL_SOFT_buffer_samples enumsChris Robinson2012-01-102-65/+65
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