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* Make the modulator effect multichannelChris Robinson2016-01-291-22/+41
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* Use separate modulator functions only for the waveformChris Robinson2016-01-291-56/+42
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* Convert the equalizer effect to multichannelChris Robinson2016-01-291-33/+90
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* Make the compressor effect multichannelChris Robinson2016-01-281-25/+58
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* Mix to multichannel for effectsChris Robinson2016-01-284-58/+258
| | | | | | 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 Robinson2016-01-2712-36/+42
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* Avoid passing the device to SetChannelMapChris Robinson2016-01-271-24/+31
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* Separate calculating ambisonic coefficients from the panning gainsChris Robinson2016-01-2513-73/+98
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* Don't write one byte at a time for the wave writer on big endianChris Robinson2016-01-251-17/+19
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* Replace multiple fputc calls with one fwriteChris Robinson2016-01-231-6/+4
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* Reorder filterstate propertiesChris Robinson2016-01-232-19/+18
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* Avoid using scandir/alphasortChris Robinson2016-01-211-23/+46
| | | | | They require POSIX 2008, which is a bit too "new" for my current liking. We can do well enough with opendir/readdir/closedir and qsort.
* Fix for systems that don't have strnlenChris Robinson2016-01-211-4/+14
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* Improve the OSS enumeration codeChris Robinson2016-01-211-63/+71
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* Merge branch 'master' of github.com:/kcat/openal-softChris Robinson2015-12-191-27/+206
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| * Added a check for an empty pathKeegan Drake H.P2015-12-171-2/+2
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| * Add device enumeration support for OSSv4Keegan Drake H.P2015-12-171-27/+206
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* | Don't overwrite the output with the dedicated effectChris Robinson2015-12-121-1/+1
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* Add includes to work around broken GNU headers and POSIX-2001Chris Robinson2015-12-112-0/+2
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* Fix slashes on the local path on WindowsChris Robinson2015-12-081-0/+3
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* Use the ALSOFT_LOCAL_PATH env var instead of the CWDChris Robinson2015-12-081-4/+18
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* Use %s.mhr for the default hrtf table listChris Robinson2015-12-051-1/+1
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* Support %s as a string matching markerChris Robinson2015-12-051-1/+94
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* Reformat Windows device name dressingChris Robinson2015-12-043-29/+27
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* Better handle inexact ALSA period countsChris Robinson2015-11-231-1/+4
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* Fix the allow-resampler check for ALSAChris Robinson2015-11-231-1/+1
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* Replace deprecated Carbon API calls with modern AudioComponent APIs in the ↵Alex Szpakowski2015-11-131-18/+24
| | | | | | | | CoreAudio backend. This prevents a deprecation notice from being output to stdout when alcOpenDevice is called in Mac OS X 10.11. The new API calls require Mac OS X 10.6 or newer.
* Use doubles for the constructed listener matrixChris Robinson2015-11-112-45/+87
| | | | | | 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.
* Move the bsincTable to a separate fileChris Robinson2015-11-102-988/+981
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* Update the bsinc tableChris Robinson2015-11-102-969/+972
| | | | Precision is increased to cover the full 32-bit float range.
* Add options to disable Pulse's and ALSA's resamplersChris Robinson2015-11-072-3/+7
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* Simplify reverb panning rotationsChris Robinson2015-11-071-4/+4
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* Remove a const to silence some warningsChris Robinson2015-11-061-1/+1
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* Use more accurate floating point literalsChris Robinson2015-11-063-41/+41
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* Add another cast for MSVCChris Robinson2015-11-061-1/+1
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* Use ALuint instead of size_t for a loop iteratorChris Robinson2015-11-061-2/+3
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* Add a cast to silence an MSVC warningChris Robinson2015-11-061-2/+2
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* Use a more appropriate type in MatchFilterChris Robinson2015-11-061-2/+2
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* Cast a double->float return to silence MSVCChris Robinson2015-11-061-1/+1
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* Pan each early/late delay line separatelyChris Robinson2015-11-051-56/+70
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* Manually inline and condense the bsinc resamplerChris Robinson2015-11-052-80/+63
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* Remove an unneeded memset that was overwriting memoryChris Robinson2015-11-051-2/+0
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* Implement a band-limited sinc resamplerChris Robinson2015-11-058-21/+1211
| | | | | | | | 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.
* Change the Kaiser rippling limit to -60dBChris Robinson2015-11-041-2/+2
| | | | | This improves the transition cutoff, shortening its width and reducing the amount of error.
* Replace the Lanczos window with Kaiser for the sinc resamplerChris Robinson2015-11-041-17/+87
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* Pass in the Q parameter for setting the filter parametersChris Robinson2015-11-015-37/+60
| | | | Also better handle the peaking filter gain.
* Use modff to split the modulation delay componentsChris Robinson2015-10-311-4/+3
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* Mix reverb to output in the inner loopChris Robinson2015-10-291-41/+39
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* Include the echo's attenuation of the late reverb in the late reverb gainChris Robinson2015-10-291-24/+29
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* Do up to 256 samples per reverb inner loop iterationChris Robinson2015-10-291-2/+3
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