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* 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 Robinson2016-03-171-0/+3
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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 Robinson2016-02-142-9/+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 Robinson2016-02-142-9/+14
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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 Robinson2016-01-311-1/+0
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* Mix to multichannel for effectsChris Robinson2016-01-282-6/+23
| | | | | | 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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* Reorder filterstate propertiesChris Robinson2016-01-231-7/+8
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* Inline a couple filterstate methodsChris Robinson2016-01-231-2/+25
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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 Robinson2015-11-012-20/+36
| | | | 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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* Rename ALC_NUM_HRTF_SPECIFIER_SOFT to ALC_NUM_HRTF_SPECIFIERS_SOFTChris Robinson2015-10-261-1/+1
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* Update filter histories even when they're not usedChris Robinson2015-10-241-0/+2
| | | | | | 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 Robinson2015-10-202-387/+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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* 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 Robinson2015-10-152-0/+5
| | | | | | 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.
* Replace the sinc6 resampler with sinc8, and make SSE versionsChris Robinson2015-10-111-4/+5
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* Move the FIR4 from SSE2 to SSE3Chris Robinson2015-10-111-2/+3
| | | | | SSE3 can avoid the slow _MM_TRANSPOSE_PS4 call thanks to the inclusion of horizontal adds.
* Allow apps to request a specific HRTFChris Robinson2015-10-071-2/+3
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* Use the enumerated HRTF list for selecting an HRTFChris Robinson2015-10-061-0/+1
| | | | Also report the proper specifier of the one currently in use.
* Enumerate and list HRTFs per-deviceChris Robinson2015-10-061-0/+1
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* Add a function to get a list of data filesChris Robinson2015-10-031-0/+2
| | | | | | The method takes a marked-up filename (e.g. may include %r for a sample rate, %% for %, etc), and returns a vector of strings of found filenames that match. It will search the CWD, the local, and global data directories, in that order.
* Add methods to enumerate and query device HRTFsChris Robinson2015-10-031-0/+4
| | | | Currently just returns a dummy entry.
* Move the resampler stuff to mixer.c where it's usedChris Robinson2015-10-012-14/+0
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* Implement a 6-point sinc-lanczos filterChris Robinson2015-09-292-2/+12
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* Replace the cubic resampler with a 4-point sinc/lanczos filterChris Robinson2015-09-272-4/+4
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* Don't keep selecting the mixer to useChris Robinson2015-09-272-3/+1
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* Increase the max pitch to 255Chris Robinson2015-09-261-1/+1
| | | | | | | Note that this is the multiple above the device sample rate, rather than the source property limit. It could theoretically be increased to 511 by testing against UINT_MAX instead of INT_MAX, since the increment and positions are using unsigned integers. I'm just being paranoid about overflows.
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