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* Avoid storing the device frequency in the distortion stateChris Robinson2013-05-211-7/+5
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* The effect state being destroyed can't be NULLChris Robinson2013-05-214-28/+22
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* Rename some inheritance macrosChris Robinson2013-05-218-73/+73
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* Pre-scale the chorus and flanger delays to be in samplesChris Robinson2013-05-212-36/+22
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* Avoid storing some chorus and flanger properties in the effect stateChris Robinson2013-05-212-52/+32
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* Process 64 samples at a time for chorus and flanger effectsChris Robinson2013-05-202-96/+130
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* Process 64 samples at a time for some effectsChris Robinson2013-05-205-117/+201
| | | | | This should help with the non-interleaved samples of the output, and allow skipping channels that don't contribute to the output.
* Split flanger delay calculation into separate methodsChris Robinson2013-05-201-69/+60
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* Split chorus delay calculation into separate methodsChris Robinson2013-05-201-37/+44
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* Use some macros to help with deriving typesChris Robinson2013-05-208-93/+81
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* Clear all the old channel gains to 0 before setting themChris Robinson2013-05-202-1/+5
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* Fix a potential leak when mmdevapi fails to openChris Robinson2013-05-191-0/+3
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* Avoid recalculating the same coefficientChris Robinson2013-05-191-11/+9
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* Implement distortion and equalizer effectsChris Robinson2013-05-183-3/+820
| | | | Code provided by Mike Gorchak
* Modulator fixesChris Robinson2013-05-181-6/+5
| | | | | | | | Modulator functions return 0..1 instead of -1..+1 Apply high-pass filter before modulation Increase fractional precision to 24 bits from 16 Thanks to Mike Gorchak
* Fix overflow issues when converting float to intChris Robinson2013-04-221-7/+10
| | | | | | | | 32-bit floats can only reach between -16777215...+16777215 (25 bits) before it starts losing whole-number accuracy. Some compiler optimizations may break trying to multiply a float by 2147483647.0 when the result is given right to a float parameter, causing it instead to multiply by the nearest representable float value, 2147483648.0.
* Add a missing cast to silence MSVCChris Robinson2013-03-221-1/+1
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* Report the actual ALSA error if setting the buffer metrics failsChris Robinson2013-03-181-2/+2
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* Use less math to clamp floats to -1...+1Chris Robinson2013-03-171-3/+3
| | | | | The previous code could have issues as precision lowers. This should hopefully work better while only using one if check instead of two.
* Use force_align_arg_pointer for QSAChris Robinson2013-03-151-0/+8
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* Add a QSA backend for QNXChris Robinson2013-03-142-0/+1173
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* Remove unnecessary NULL checksChris Robinson2013-03-132-22/+14
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* Add Chorus and Flanger effectsChris Robinson2013-03-133-2/+806
| | | | Code provided by Mike Gorchak
* Move the effect-specific get/set methods to where the effect is implementedChris Robinson2013-03-134-0/+776
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* Build an appropriate channel map for PulseAudioChris Robinson2013-02-081-2/+28
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* Destroy the ring buffer when closing down a portaudio deviceChris Robinson2013-02-011-0/+3
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* Avoid an underflow in the SSE mixers if BufferSize is less than 4Chris Robinson2012-12-241-2/+2
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* Filter unneeded thread messages with MMDevApiChris Robinson2012-12-241-1/+3
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* Add support for OpenSL ES 1.0.1 on AndroidChris Robinson2012-12-121-1/+8
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* Trace the list of supported backendsChris Robinson2012-12-041-0/+7
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* Use +/-90 degrees for stereo sources with non-HRTF stereo output.Chris Robinson2012-12-041-1/+9
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* Add support for float32 output to CoreAudioChris Robinson2012-12-031-10/+13
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* Use the correct multiplier for int32 output with CoreAudioChris Robinson2012-12-031-2/+2
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* Lock the device before calling aluHandleDisconnectChris Robinson2012-12-029-2/+30
| | | | | | PulseAudio causes an assert if being relocked inside a callback on the worker thread, where aluHandleDisconnect is called. We can assume it's already locked there, so just make sure the device is locked before being calling it.
* Trace the opened device name in alcCaptureOpenDeviceChris Robinson2012-12-021-1/+1
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* Use WARN when implicitly deleting resources with the contextChris Robinson2012-12-021-2/+2
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* Explicitly check for malloc.hChris Robinson2012-11-251-1/+1
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* malloc.h isn't standard, and is only needed for _aligned_mallocChris Robinson2012-11-251-1/+3
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* Avoid locking PulseAudio's mainloop when starting and stopping captureChris Robinson2012-11-191-4/+0
| | | | | The mainloop is already locked by the caller, and double-locking interferes with the subsequent wait.
* Fix off-by-one error in the makehrtf dither and add a resample optionChris Robinson2012-11-181-831/+829
| | | | Also fixes some size_t type warnings
* Avoid storing the SSE state if it's not supportedChris Robinson2012-11-131-1/+4
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* Include standard headers before conditional onesChris Robinson2012-11-041-7/+6
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* Use the stored buffer metrics for the mmdevapi buffer sizeChris Robinson2012-11-041-9/+1
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* Add rudimentary latency tracking for mmdevapiChris Robinson2012-11-041-1/+16
| | | | | | | | This won't be as granular as it could be, since it only updates when the wakeup event trips (which may or may not happen more often than OpenAL's mix updates). A more correct method would be to query GetCurrentPadding directly, but that would require sending a message to the processing thread and waiting for a reply, since we can't guarantee COM on the calling thread.
* Use an array to specify the offset for each channel of the device bufferChris Robinson2012-11-042-60/+73
| | | | This effectively inverts the DevChannels array
* Rename some struct fields for DirectSoundChris Robinson2012-11-041-48/+48
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* Avoid recreating the DSound primary bufferChris Robinson2012-11-041-1/+1
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* Make a decimal value a float typeChris Robinson2012-10-311-1/+1
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* Finalize AL_SOFT_source_latencyChris Robinson2012-10-311-1/+1
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* Add a couple comments to the codeChris Robinson2012-10-261-0/+2
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