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* Add a cmake option to embed the HRTF dataChris Robinson2016-07-125-7/+69
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* Store the voice output buffers separate from the paramsChris Robinson2016-07-114-40/+44
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* Update comment about the source radius calculationsChris Robinson2016-07-101-9/+18
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* Fix a direct access to a UIntMapChris Robinson2016-07-091-2/+2
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* Reorder some source fieldsChris Robinson2016-07-072-24/+27
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* Avoid standard malloc/free for Hrtf allocationChris Robinson2016-07-071-3/+4
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* Avoid function calls to get the HRTF sample rate and IR sizeChris Robinson2016-07-076-36/+21
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* Avoid using memcpy to copy a single structChris Robinson2016-07-063-21/+19
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* Remove a slightly outdated commentChris Robinson2016-07-061-1/+0
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* Remove the VirtOut buffer aliasChris Robinson2016-07-053-30/+24
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* Use separate arrays for UIntMap keys and valuesChris Robinson2016-07-048-58/+87
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* Use al_malloc/al_free for the UIntMap arrayChris Robinson2016-07-042-9/+17
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* Ensure voices has been updated once before mixing themChris Robinson2016-06-162-2/+10
| | | | | | | | Sometimes the mixer is temporarily prevented from applying updates, when multiple sources need to be updated simultaneously for example, but does not prevent mixing. If the mixer runs during that time and a voice was just started, it would've mixed the voice without any internal properties being set for it.
* Rename MaxNoOfSources for consistencyChris Robinson2016-06-082-16/+16
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* Remove an IN_IDE_PARSER hackChris Robinson2016-06-071-6/+0
| | | | | | Not all IDE parsers necessarily choke on restrict, and even those that do can probably have their own configuration to define macros that can hide the parsing errors caused by it.
* Look in the executable's dir for another config fileChris Robinson2016-06-043-0/+133
| | | | | On Windows it'll look for alsoft.ini, and elsewhere is alsoft.conf. This applies after the user-local settings and before ALSOFT_CONF.
* Only define 8 HRTF filter states and params for the deviceChris Robinson2016-06-041-2/+2
| | | | | It will only be used with a cube channel setup, so there's no need to have one for every possible output channel.
* Add property queries to get the device latency with the clockChris Robinson2016-06-032-11/+34
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* Improve the filter processing functionChris Robinson2016-06-031-2/+35
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* Remove some unnecessary volatile keywordsChris Robinson2016-06-032-33/+33
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* Implement a Neon-enhanced MixRowChris Robinson2016-06-013-0/+31
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* Make a function staticChris Robinson2016-06-012-7/+7
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* Use a macro to specify the ambisonic periphonic channel maskChris Robinson2016-06-013-4/+13
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* Use SSE for applying the HQ B-Format decoder matricesChris Robinson2016-05-315-27/+82
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* Don't access the band splitter fields in the processing loopsChris Robinson2016-05-311-9/+16
| | | | | | perf shows a 5% drop in relative execution time on the alffplay example with an audio-only file (20% to 15%). Kinda figured the optimizer would handle it better, but I guess not.
* Don't try to emulate almtx_timedlockChris Robinson2016-05-301-43/+12
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* Clean up a couple variable names and declarationsChris Robinson2016-05-302-7/+6
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* Remove unnecessary VECTOR_INSERTChris Robinson2016-05-302-54/+30
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* Hold the effectslot map lock while handling itChris Robinson2016-05-293-2/+38
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* Use a linked list for active effect slotsChris Robinson2016-05-295-60/+59
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* Avoid the mixer lock when getting the plain source offsetChris Robinson2016-05-281-11/+15
| | | | i.e. without the latency
* Avoid an explicit mixer lock for getting the source offset and latencyChris Robinson2016-05-281-27/+72
| | | | | The only mixer locking involved is with the backend, as determined by it's ability to get the device clock and latency atomically.
* Change the backend getLatency method to return the clock time tooChris Robinson2016-05-2815-53/+119
| | | | | | This will also allow backends to better synchronize the tracked clock time with the device output latency, without necessarily needing to lock if the backend API can allow for it.
* Use the backend lock for the effectstate's deviceUpdate callChris Robinson2016-05-271-8/+4
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* Use a specific lock for the backend's stop/reset/play callsChris Robinson2016-05-272-30/+46
| | | | | This helps protect against the device changing unexpectedly from multiple threads, instead of using the global list/library lock.
* Remove a couple unneeded functionsChris Robinson2016-05-253-24/+7
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* Increment the device's mix count closer to the mixing loopsChris Robinson2016-05-231-3/+2
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* Properly pluralize some messagesChris Robinson2016-05-222-8/+14
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* Improve locking for device attribute queriesChris Robinson2016-05-221-8/+28
| | | | | Avoids the backend device lock, instead using the list lock to prevent the device from changing while being queried, and adds some missing locks.
* Avoid using realloc in a number of placesChris Robinson2016-05-219-32/+53
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* Make the source position calues atomicChris Robinson2016-05-194-30/+31
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* Copy the source's Looping property into the voiceChris Robinson2016-05-181-0/+2
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* Avoid redundantly storing distance model settingsChris Robinson2016-05-176-33/+11
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* Update the right effect state when the device is resetChris Robinson2016-05-171-2/+2
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* Don't assume the "real" output buffer follows the dry bufferChris Robinson2016-05-171-15/+44
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* Improve reverb panning gains for "3D" output.Chris Robinson2016-05-171-8/+8
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* Ignore the listening angle for the wet path sound conesChris Robinson2016-05-161-32/+51
| | | | | | | Since the wet path is essentially the room response to a sound, the direction of the sound to the listener doesn't change the amount of energy the room receives. Instead, the surface area defined by the cones dictate the volume the room gets for the sound.
* Use floats for the listener transformsChris Robinson2016-05-164-92/+29
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* Don't store the source's update method with the voiceChris Robinson2016-05-164-48/+42
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* Avoid separate updates to sources that should apply togetherChris Robinson2016-05-155-15/+37
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