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* Pas the output device channel count to ALeffectState::processChris Robinson2014-11-072-2/+6
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* Rename speakers to channels, and remove an old incorrect commentChris Robinson2014-11-071-4/+2
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* Use a separate macro for the max output channel countChris Robinson2014-11-072-12/+11
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* Fix 5.1 surround soundChris Robinson2014-11-071-2/+2
| | | | | | | | | | | | | Apparently, 5.1 surround sound is supposed to use the "side" channels, not the back channels, and we've been wrong this whole time. That means the "5.1 Side" is actually the correct 5.1 setup, and using the back channels is anomalous. Additionally, this means the 5.1 buffer format should also use the the side channels instead of the back channels. A final note: the 5.1 mixing coefficients are changed so both use the original 5.1 surround sound set (with the surround channels at +/-110 degrees). So the only difference now between 5.1 "side" and 5.1 "back" is the channel labels.
* Remove the channel name from ChannelConfigChris Robinson2014-11-051-1/+0
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* Use a method to set omni-directional channel gainsChris Robinson2014-11-041-14/+7
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* Support B-Format source rotation with AL_ORIENTATIONChris Robinson2014-10-312-3/+5
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* Rename the source's Orientation to DirectionChris Robinson2014-10-311-1/+1
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* Add preliminary AL_EXT_BFORMAT supportChris Robinson2014-10-313-1/+14
| | | | | Currently missing the AL_ORIENTATION source property. Gain stepping also does not work.
* Make alcSuspendContext and alcProcessContext batch updatesChris Robinson2014-10-121-0/+3
| | | | | | | | | | This behavior better matches Creative's hardware drivers and Rapture3D's OpenAL driver. A compatibility environment variable is provided to restore the old no-op behavior for any app that behaves badly from this change (set __ALSOFT_SUSPEND_CONTEXT to "ignore"). If too many apps have a problem with this, the default behavior may need to be changed to ignore, with the env var providing an option to defer/batch instead.
* Add a helper to search for a channel index by nameChris Robinson2014-10-021-0/+17
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* Store default speaker configurations in a structChris Robinson2014-10-021-11/+14
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* Make ComputeAngleGains use ComputeDirectionalGainsChris Robinson2014-10-022-5/+6
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* Don't use ComputeAngleGains for SetGainsChris Robinson2014-10-021-1/+5
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* Use an ambisonics-based panning methodChris Robinson2014-09-302-0/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For mono sources, third-order ambisonics is utilized to generate panning gains. The general idea is that a panned mono sound can be encoded into b-format ambisonics as: w[i] = sample[i] * 0.7071; x[i] = sample[i] * dir[0]; y[i] = sample[i] * dir[1]; ... and subsequently rendered using: output[chan][i] = w[i] * w_coeffs[chan] + x[i] * x_coeffs[chan] + y[i] * y_coeffs[chan] + ...; By reordering the math, channel gains can be generated by doing: gain[chan] = 0.7071 * w_coeffs[chan] + dir[0] * x_coeffs[chan] + dir[1] * y_coeffs[chan] + ...; which then get applied as normal: output[chan][i] = sample[i] * gain[chan]; One of the reasons to use ambisonics for panning is that it provides arguably better reproduction for sounds emanating from between two speakers. As well, this makes it easier to pan in all 3 dimensions, with for instance a "3D7.1" or 8-channel cube speaker configuration by simply providing the necessary coefficients (this will need some work since some methods still use angle-based panpot, particularly multi-channel sources). Unfortunately, the math to reliably generate the coefficients for a given speaker configuration is too costly to do at run-time. They have to be pre- generated based on a pre-specified speaker arangement, which means the config options for tweaking speaker angles are no longer supportable. Eventually I hope to provide config options for custom coefficients, which can either be generated and written in manually, or via alsoft-config from user-specified speaker positions. The current default set of coefficients were generated using the MATLAB scripts (compatible with GNU Octave) from the excellent Ambisonic Decoder Toolbox, at https://bitbucket.org/ambidecodertoolbox/adt/
* Combine some fields into a structChris Robinson2014-09-101-3/+6
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* Invert the ChannelOffsets arrayChris Robinson2014-09-101-1/+2
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* Remove the GetLatency method from the old BackendFuncsChris Robinson2014-09-081-4/+0
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* Convert the winmm backend to the new backend APIChris Robinson2014-09-081-3/+0
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* Make the fontsound's buffer and link fields atomicChris Robinson2014-09-031-2/+3
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* Make the buffer's pack and unpack properties atomicChris Robinson2014-09-031-2/+2
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* Remove a couple unnecessary typedefsChris Robinson2014-08-241-3/+0
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* Convert the wave writer backend to the new APIChris Robinson2014-08-241-3/+0
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* Use al_malloc/al_free for default allocatorsChris Robinson2014-08-241-2/+2
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* Rename activesource to voiceChris Robinson2014-08-213-10/+10
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* Use an array of objects for active sources instead of pointersChris Robinson2014-08-211-1/+1
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* Use a NULL source for inactive activesourcesChris Robinson2014-08-212-5/+12
| | | | Also only access the activesource's source field once per update.
* ALC_SOFT_pause_device is finishedChris Robinson2014-08-121-10/+0
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* Use atomics for the device and context list headsChris Robinson2014-08-011-2/+2
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* Make the source's buffer queue head and current queue item atomicChris Robinson2014-07-311-5/+5
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* Use generic atomics in more placesChris Robinson2014-07-222-3/+3
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* Add macros for generic atomic functionalityChris Robinson2014-07-222-2/+2
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* Make some functions staticChris Robinson2014-07-201-2/+0
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* Add a source radius property that determines the directionality of a soundChris Robinson2014-07-111-0/+2
| | | | | | | | | At 0 distance from the listener, the sound is omni-directional. As the source and listener become 'radius' units apart, the sound becomes more directional. With HRTF, an omni-directional sound is handled using 0-delay, pass-through filter coefficients, which is blended with the real delay and coefficients as needed to become more directional.
* Store 4 modulators per map entryChris Robinson2014-07-061-0/+1
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* Regroup and reorganize some macrosChris Robinson2014-07-061-40/+57
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* Don't require pre-declaring vector typesChris Robinson2014-07-061-5/+1
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* Make some more functions staticChris Robinson2014-07-051-5/+0
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* Use a helper function to check valid MIDI controller inputsChris Robinson2014-07-041-0/+14
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* AL_SOFT_MSADPCM is functionally completeChris Robinson2014-07-031-6/+0
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* Standardize some New/Delete methodsChris Robinson2014-06-302-3/+3
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* Remove an unused macroChris Robinson2014-06-291-1/+0
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* Load soundfont samples into an ALbufferChris Robinson2014-06-293-12/+5
| | | | | Also remove ALsoundfont's now-unneeded sample storage functions and struct fields.
* Store and use an ALbuffer for samples in an ALfontsoundChris Robinson2014-06-291-0/+2
| | | | | | | | The fontsound still maintains its own start, end, and loop offsets, so that the same buffer may be shared between multiple/all fontsounds. Ideally a single buffer should be used for all fontsounds to avoid memory fragmentation and help CPU caching, although higher quality soundfonts may need more memory than a single buffer can hold.
* Get the mixer and resampler functions when neededChris Robinson2014-06-131-4/+1
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* Combine the direct and send mixersChris Robinson2014-06-132-8/+5
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* Combine some dry and wet path typesChris Robinson2014-06-132-17/+8
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* Pre-apply the crossfeed filter gain to the input sample coefficientsChris Robinson2014-06-121-19/+10
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* Use floats for the BS2B filterChris Robinson2014-06-111-10/+10
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* Make bs2b_cross_feed inlineChris Robinson2014-06-101-2/+33
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