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* Rename UpdateFreq device field to UpdateSizeChris Robinson2008-02-125-20/+20
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* Pretend DSound buffer fragment size is 1/4th the total buffer sizeChris Robinson2008-02-121-2/+7
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* Make the backend list staticChris Robinson2008-02-111-1/+1
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* Call InitAL at the start of some more ALC functionsChris Robinson2008-02-111-0/+8
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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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* Remove explicit dependancy on ole32 and unused dxguidChris Robinson2008-02-081-9/+1
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* Enumerate DirectSound devicesChris Robinson2008-02-081-9/+39
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* Remove unneeded device struct memberChris Robinson2008-02-081-2/+0
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* Prevent overflow of the device listsChris Robinson2008-02-081-6/+21
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* Use all capture devices listed by ALSA, not just the first on each cardChris Robinson2008-02-081-12/+35
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* Don't remove the devices from the device list on unexpected shutdownChris Robinson2008-02-081-7/+3
| | | | The close functions will remove it for us
* Set the done flag immediately when entering the initializationChris Robinson2008-02-061-1/+2
| | | | | To prevent two threads from initializing at the same time (not that it's likely to happen at this point).
* Add an option for duplicating stereo sources on the back speakersChris Robinson2008-02-062-7/+24
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* Fix race condition when starting winmm message poll threadChris Robinson2008-02-031-9/+22
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* Use WAVEFORMATEXTENSIBLE for multichannel dsound output, and don't create a ↵Chris Robinson2008-02-011-18/+51
| | | | primary buffer
* 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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* Fix availibility amount calculationChris Robinson2008-01-261-2/+5
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* aluBytesFromFormat returns bytes, not bitsChris Robinson2008-01-261-5/+5
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* Update frame size after detecting the directsound output formatChris Robinson2008-01-251-0/+2
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* Remove unnecessary Channels fieldChris Robinson2008-01-255-54/+20
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* Set the output format according to the speaker setup reported by directsoundChris Robinson2008-01-251-8/+54
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* Use both write pointers from the directsound buffer lockChris Robinson2008-01-251-8/+10
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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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* Don't use a multiple lists for extensionsChris Robinson2008-01-191-20/+20
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* Add an option for setting the max number of sourcesChris Robinson2008-01-191-1/+3
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* Remove duplication of setting the max source countChris Robinson2008-01-195-6/+2
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* Use less ambiguous config file namesChris Robinson2008-01-191-2/+17
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* Remove some unnecessary duplicate math, which was making long linesChris Robinson2008-01-191-67/+56
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* Remove some branchesChris Robinson2008-01-181-75/+25
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* Implement AL_EFFECT_REVERBChris Robinson2008-01-181-19/+98
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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. :)
* Remove duplicated source freeing codeChris Robinson2008-01-181-20/+0
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* Buffer size fixes. Partially reverts the ALSA buffer size "fix"Chris Robinson2008-01-182-11/+17
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* Don't dereference ALContext if there's no context yetChris Robinson2008-01-171-2/+2
| | | | Patch by Evgeny A. Marchenko
* Add missing config.h includesChris Robinson2008-01-167-2/+14
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* Don't include alAuxEffectSlot.h in alSource.hChris Robinson2008-01-162-0/+2
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* Make sure sources are deleted with the contextChris Robinson2008-01-161-0/+1
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* Don't clamp wet gain if there's no send slot, and move slot gain calculationChris Robinson2008-01-161-9/+12
| | | | To remove an extra if check
* Store a reference to the effect slot in a source's send, not a copyChris Robinson2008-01-161-11/+13
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* Remove unneeded variablesChris Robinson2008-01-151-38/+28
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* Use acosf when availableChris Robinson2008-01-151-1/+8
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* Use the previous low-pass filter again, as it seems to match the intended ↵Chris Robinson2008-01-151-6/+14
| | | | output better
* Store effect slots in the contextChris Robinson2008-01-151-0/+2
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* Add support for AL_LOKI_quadriphonicChris Robinson2008-01-142-1/+5
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* Reduce indentationChris Robinson2008-01-141-6/+4
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