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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 Robinson2016-03-294-5/+6
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* Include any first-order scaling in the FOAOut coefficientsChris Robinson2016-03-251-2/+0
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* Implement AL_EXT_STEREO_ANGLES supportChris Robinson2016-03-252-0/+41
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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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* Store the effect's output buffer in the effect stateChris Robinson2016-03-172-0/+5
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* Rename the BFormat channels to AuxChris Robinson2016-03-161-4/+4
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* Add config options to enable the hq ambisonic decoderChris Robinson2016-03-161-2/+1
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* Add a dual-band ambisonic decoderChris Robinson2016-03-152-1/+5
| | | | | | | | | | 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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* Use the real output's left and right channels with HRTFChris Robinson2016-03-111-2/+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 Robinson2016-02-143-10/+11
| | | | | 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.
* Calculate channel gain stepping just before mixingChris Robinson2016-02-143-16/+16
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* Rename ComputeBFormatGains to ComputeFirstOrderGainsChris Robinson2016-01-311-5/+5
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* Make the source's buffer queue a singly-linked listChris Robinson2016-01-312-31/+27
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* Mix to multichannel for effectsChris Robinson2016-01-283-6/+25
| | | | | | This mixes to a 4-channel first-order ambisonics buffer. With ACN ordering and N3D scaling, this makes it easy to remain compatible with effects that only care about mono input since channel 0 is an unattenuated mono signal.
* Pass a pointer to the input samples array for effect processingChris Robinson2016-01-271-2/+2
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* Separate calculating ambisonic coefficients from the panning gainsChris Robinson2016-01-252-16/+36
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* Move a couple extern inline declarations to the othersChris Robinson2016-01-231-3/+2
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* Reorder filterstate propertiesChris Robinson2016-01-232-49/+51
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* Inline a couple filterstate methodsChris Robinson2016-01-232-27/+27
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* Use doubles for the constructed listener matrixChris Robinson2015-11-112-14/+39
| | | | | | This helps the stability of transforms to local space for sources that are at or near the listener. With a single-precision matrix, even FLT_EPSILON might not be enough to detect matching positions.
* Implement a band-limited sinc resamplerChris Robinson2015-11-052-4/+32
| | | | | | | | This is essentially a 12-point sinc resampler, unless it's resampling to a rate higher than the output, at which point it will vary between 12 and 24 points and do anti-aliasing to avoid/reduce frequencies going over nyquist. Code provided by Christopher Fitzgerald.
* Pass in the Q parameter for setting the filter parametersChris Robinson2015-11-013-27/+41
| | | | Also better handle the peaking filter gain.
* Remove an unused struct fieldChris Robinson2015-11-011-1/+0
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* Fix a commentChris Robinson2015-11-011-1/+1
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* Finalize ALC_SOFT_HRTFChris Robinson2015-10-281-22/+0
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* Always update all reverb propertiesChris Robinson2015-10-281-0/+14
| | | | | | The EAX-only effect properties will be set to compatible defaults when standard reverb is set, and the EAX-only effects will be skipped during sample processing.
* Rename ALC_NUM_HRTF_SPECIFIER_SOFT to ALC_NUM_HRTF_SPECIFIERS_SOFTChris Robinson2015-10-261-1/+1
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* Lock the source queue for writing when updating the playback offsetChris Robinson2015-10-241-8/+8
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* Fix usage of modfChris Robinson2015-10-241-2/+2
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* Update filter histories even when they're not usedChris Robinson2015-10-242-0/+20
| | | | | | If the filter properties are continually updated, and the HF or LF gain goes from <1, to 1, and later back to <1, the history shouldn't hold stale values from before it was at 1.
* Use one send gain per buffer channelChris Robinson2015-10-231-1/+1
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* Convert the PortAudio backend to the new backend APIChris Robinson2015-10-221-3/+0
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* Remove the MIDI codeChris Robinson2015-10-207-2508/+0
| | | | | | | The extension's not going anywhere, and it can't do anything fluidsynth can't. The code maintenance and bloat is not worth keeping around, and ideally the AL API would be able to facilitate MIDI-like behavior anyway (envelopes, start-at- time, etc).
* Remove unused channel labelsChris Robinson2015-10-181-9/+0
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* Include the fractional part with the source sample/sec offsetChris Robinson2015-10-161-17/+18
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* Use a constant value for the post-position paddingChris Robinson2015-10-152-3/+6
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* Store the source's previous samples with the voiceChris Robinson2015-10-153-0/+17
| | | | | | This helps avoid different results when looping is toggled within a couple samples of the loop point, or when a processed buffer is removed while the source is only a couple samples into the next buffer.
* Properly limit the calculated source offset componentsChris Robinson2015-10-141-4/+8
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* Shut GCC upChris Robinson2015-10-131-1/+1
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