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* Add the Echo effectChris Robinson2009-04-123-0/+60
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* Pay attention to the MAX_SENDS valueChris Robinson2009-04-111-2/+1
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* Make auxiliary effect slot count configurableChris Robinson2009-04-111-0/+2
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* Move the WetBuffer into the effect slot objectChris Robinson2009-04-112-0/+4
| | | | This should make it easier to support multiple slots
* Remove the SDL backendChris Robinson2009-03-101-1/+0
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* Add a PortAudio backendChris Robinson2009-03-101-0/+1
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* Use a matrix for up- and down-mixing channelsChris Robinson2009-01-251-0/+2
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* 6.1 uses front- and back-center, not left- and right-back channelsChris Robinson2009-01-241-1/+2
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* Reimplement panning using lookup tables, based on a patch by Christian BorssChris Robinson2009-01-242-0/+7
| | | | | | This allows speaker positions to be specified by discrete angles around the listener, providing more flexibility and configurability in placement. Additional patches to take advantage of this are forthcoming.
* Apply the dry filter to multi-channel sourcesChris Robinson2008-12-101-1/+2
| | | | Unlike mono sources, they use 2 chained one-pole filters instead of 4
* Add an SDL backendChris Robinson2008-12-071-0/+1
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* Implement AL_EXTX_source_distance_modelChris Robinson2008-11-251-0/+1
| | | | | As with other EXTX extensions, this is subject to change and removal as the spec gets worked on
* Remove unneeded macroChris Robinson2008-11-161-1/+0
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* Implement a new reverb effectChris Robinson2008-11-162-9/+28
| | | | Code created and graciously provided by Christopher Fitzgerald
* Add an option to disable specific EFX effect typesChris Robinson2008-11-141-0/+7
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* Don't ramp gains when starting a sound from the beginningChris Robinson2008-11-131-0/+1
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* Add initial AL_EXTX_buffer_sub_data supportChris Robinson2008-11-111-0/+2
| | | | | | | | Note that this is an in-development extension, as noted by the EXTX moniker instead of EXT. It's behavior is subject to change, and the extension string will be removed (replaced with the official string once it's finalized). Developers are discouraged from using this in production code, though feel free to play around with it.
* Remove another unused source memberChris Robinson2008-10-101-2/+0
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* Remove unneeded source member variableChris Robinson2008-10-091-1/+0
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* Commit missing changesChris Robinson2008-10-091-1/+1
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* Use a new low-pass filter, based on the I3DL2 specChris Robinson2008-10-022-6/+3
| | | | Many thanks to Christopher Fitzgerald, for helping with it
* Air absorption factor is applied to the dB value, not linear gainChris Robinson2008-09-221-0/+1
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* Use a 12dB/oct rolloff instead of 24 for the lowpass filterChris Robinson2008-09-131-4/+2
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* Add a Solaris playback backendChris Robinson2008-09-071-0/+1
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* Don't export extension function symbols from the libChris Robinson2008-09-063-39/+39
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* Remove unneeded source struct memberChris Robinson2008-08-151-1/+1
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* Ramp channel gains to remove pops and clicks from abrupt changesChris Robinson2008-08-142-0/+20
| | | | Thanks to Christopher Fitzgerald for helping me work on it
* Include fenv.h if it exists for fesetroundChris Robinson2008-08-081-0/+4
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* Use arrays instead of pointer-to-arrays for the low-pass filterChris Robinson2008-07-261-4/+4
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* Make the filter processing function inlineChris Robinson2008-07-261-1/+2
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* Implement yet another low-pass filterChris Robinson2008-07-253-2/+14
| | | | This one using the Butterworth IIR filter design
* Specify padding per buffer, and make sure it's large enough for the filter stepChris Robinson2008-07-242-0/+4
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* Implement an alternative low-pass filterChris Robinson2008-07-231-3/+0
| | | | | | | | | 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
* Store extension list with a pointer, not a per-context arrayChris Robinson2008-07-221-1/+1
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* Move (de)initialization into ALc.c and remove unneeded fileChris Robinson2008-07-171-2/+0
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* Implement doppler factor source propertyChris Robinson2008-07-151-0/+2
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* Use pthread_mutexattr_setkind_np as a fallback to set a recursive mutex typeChris Robinson2008-05-151-0/+7
| | | | Some systems (FreeBSD) don't like setting it through pthread_mutexattr_settype
* constify the pointer that holds the filenameChris Robinson2008-03-221-8/+8
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* Define _WIN32_WINNT to 0x0500 when including windows.hChris Robinson2008-03-011-0/+3
| | | | VC7 appears to require that value, or higher, set and fails otherwise
* Remove FrameSize struct memberChris Robinson2008-02-141-1/+0
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* Rename UpdateFreq device field to UpdateSizeChris Robinson2008-02-121-1/+1
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* Include alext.h instead of redefining some enumsChris Robinson2008-02-081-27/+4
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* Remove unneeded device struct memberChris Robinson2008-02-081-1/+0
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* Add an option for duplicating stereo sources on the back speakersChris Robinson2008-02-061-0/+2
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* Remove unnecessary Channels fieldChris Robinson2008-01-251-1/+0
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* Implement AL_EFFECT_REVERBChris Robinson2008-01-181-0/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Here is a quick description of how the reverb effect works: +--->---+*(4) | V new sample +-----+---+---+ | |extra|ltr|ref| <- +*(1) +-----+---+---+ (3,5)*| |*(2) +-->| V out sample 1) Apply master reverb gain to incoming sample and place it at the head of the buffer. The master reverb gainhf was already applied when the source was initially mixed. 2) Copy the delayed reflection sample to an output sample and apply the reflection gain. 3) Apply the late reverb gain to the late reverb sample 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio, and add to the late reverb. 5) Copy the late reverb sample, adding to the output sample. Then the head and sampling points are shifted forward, and done again for each new sample. The extra buffer length is determined by the Reverb Density property. A value of 0 gives a length of 0.1 seconds (long, with fairly distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos). The decay gain is calculated such that after a number of loops to satisfy the Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to the resulting output, and only getting further reduced). It is calculated as: DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength)); Things to note: Reverb Diffusion is not currently handled, nor is Decay HF Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this method likely sucks, but it's the best I can come up with before release. :)
* Don't include alAuxEffectSlot.h in alSource.hChris Robinson2008-01-161-2/+1
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* Make sure sources are deleted with the contextChris Robinson2008-01-161-0/+2
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* Keep track of references to effect slots, so they aren't deleted while in useChris Robinson2008-01-161-0/+2
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* Store a reference to the effect slot in a source's send, not a copyChris Robinson2008-01-161-1/+1
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