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* Implement AL_EXTX_source_distance_modelChris Robinson2008-11-251-1/+1
| | | | | As with other EXTX extensions, this is subject to change and removal as the spec gets worked on
* Use a better dB-to-linear gain convertionChris Robinson2008-11-161-1/+1
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* Implement a new reverb effectChris Robinson2008-11-161-59/+13
| | | | Code created and graciously provided by Christopher Fitzgerald
* Don't ramp gains when starting a sound from the beginningChris Robinson2008-11-131-3/+17
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* Include float.h if it exists, for _RC_CHOP and _MCW_RCChris Robinson2008-10-141-0/+4
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* Remove another unused source memberChris Robinson2008-10-101-1/+0
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* Use a modulo to keep the buffer position in range for looping sourcesChris Robinson2008-10-101-1/+4
| | | | | A high pitch and low buffer size can cause a lot of unnecessary iterations otherwise, that just decrement the position
* Clamp source position to the buffer size when it stopsChris Robinson2008-10-091-0/+2
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* Remove unneeded source member variableChris Robinson2008-10-091-1/+0
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* Only send one channel through the wet pathChris Robinson2008-10-091-149/+106
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* Increase max pitch to 65536Chris Robinson2008-10-091-4/+4
| | | | This should be safe now
* Simplify the lerp functionChris Robinson2008-10-091-1/+1
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* Don't apply the wet path for multi-channel buffersChris Robinson2008-10-091-10/+0
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* Skip mixing if the read position is beyond the end of the bufferChris Robinson2008-10-091-3/+8
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* The wet path should be silent if no effect is set on the slotChris Robinson2008-10-091-1/+2
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* Don't hold the whole-number position in the fractional valueChris Robinson2008-10-021-24/+19
| | | | This will help prevent overflows when the max pitch is increased
* Use a new low-pass filter, based on the I3DL2 specChris Robinson2008-10-021-47/+40
| | | | Many thanks to Christopher Fitzgerald, for helping with it
* Air absorption factor is applied to the dB value, not linear gainChris Robinson2008-09-221-13/+15
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* Fixup some source parameter calculationsChris Robinson2008-09-161-28/+49
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* Use a 12dB/oct rolloff instead of 24 for the lowpass filterChris Robinson2008-09-131-14/+10
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* Clear the end of the buffer when at the end of the queue and not loopingChris Robinson2008-09-061-0/+2
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* Remove unneeded source struct memberChris Robinson2008-08-151-4/+1
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* Overwrite the input wet sample with the outputChris Robinson2008-08-141-6/+6
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* Ramp channel gains to remove pops and clicks from abrupt changesChris Robinson2008-08-141-20/+52
| | | | Thanks to Christopher Fitzgerald for helping me work on it
* Set FPU mode to round toward zero for mixingChris Robinson2008-08-081-0/+17
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* Remove unnecessary castingChris Robinson2008-08-081-8/+16
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* Prevent a 0 or negative increment for the buffer positionChris Robinson2008-08-051-0/+2
| | | | Thanks to Christopher Fitzgerald for pointing these last two problems out
* Fix some calculations for the reverb bufferChris Robinson2008-07-261-25/+22
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* Make the filter processing function inlineChris Robinson2008-07-261-0/+36
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* Implement yet another low-pass filterChris Robinson2008-07-251-16/+9
| | | | This one using the Butterworth IIR filter design
* Specify padding per buffer, and make sure it's large enough for the filter stepChris Robinson2008-07-241-5/+5
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* Don't advertise extra samples for mixingChris Robinson2008-07-231-3/+2
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* Implement an alternative low-pass filterChris Robinson2008-07-231-36/+32
| | | | | | | | | This method samples from the buffer so that it gets a time-correct 5khz stream, which is subtracted from the original sample and has the high-frequency gain applied, then added back. A better method may be to average all the samples from the current one to the one freq/5000 away, instead of bilinear filtering the two nearest freq/5000 apart. Processing cost will need to determine its viability
* Implement doppler factor source propertyChris Robinson2008-07-151-1/+1
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* Add the reverb room rolloff to the source room rolloff, not overrideChris Robinson2008-07-151-1/+1
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* Reduce the mix buffer sizes by halfChris Robinson2008-07-081-1/+1
| | | | Nearly 3MB is a bit much. Could reduce it further, but this is good enough for now.
* Leave SourceToListener untransformed for use with untransformed velocitiesChris Robinson2008-07-031-6/+16
| | | | | Distance is also left untransformed so cone calculations with SoundToListener are correct
* Fix source calculations for AL_SOURCE_RELATIVE modeChris Robinson2008-05-181-18/+22
| | | | | | Make sure the source position and direction are properly put into listener- space before working with them, and don't calculate the listener velocity for relative coordinates
* Check the right struct member for the filter typeChris Robinson2008-04-121-2/+2
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* Fast float-to-int function is no longer neededChris Robinson2008-02-081-14/+2
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* Remove unnecessary castingChris Robinson2008-02-081-2/+2
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* Add an option for duplicating stereo sources on the back speakersChris Robinson2008-02-061-6/+17
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* Use the correct channel ordering for WindowsChris Robinson2008-01-271-0/+40
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* Fix output channel order for 6.1 and 7.1Chris Robinson2008-01-271-22/+22
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* Remove an unneceesary pointer check and decrease indentationChris Robinson2008-01-211-424/+421
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* Remove unnecessary duplicate thunk lookupsChris Robinson2008-01-211-10/+8
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* Small formatting updatesChris Robinson2008-01-201-1/+3
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* Remove duplicate functionChris Robinson2008-01-201-23/+7
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* Don't access ALSource for every sample mixChris Robinson2008-01-201-21/+24
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* Remove some unnecessary duplicate math, which was making long linesChris Robinson2008-01-191-67/+56
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