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* Add forwarding for the AL functionsChris Robinson2017-06-283-7/+19
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* Load driver dlls in the routerChris Robinson2017-06-284-3/+378
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* Start a router DLLChris Robinson2017-06-284-10/+265
| | | | | | | | | | Experimental, Windows only. This is intended as an alternative to Creative's router DLL, fixing a few issues with it (falsely reporting extensions that aren't supported, not being able to query the ALC version without a device, and not being able to use ALC extension functions). When enabled OpenAL Soft's DLL is built as soft_oal.dll, while the router is OpenAL32.dll.
* Calculate the converter stepping value using floating pointChris Robinson2017-06-281-3/+3
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* Remove the fastf2u conversion functionChris Robinson2017-06-277-18/+12
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* Use a macro to apply NFC filtered mixes instead of a loopChris Robinson2017-06-261-18/+16
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* Round the converter's stepping valueChris Robinson2017-06-261-2/+3
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* Trace the message name in the message handler loopChris Robinson2017-06-261-1/+14
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* Convert all input samples in the loopChris Robinson2017-06-261-2/+2
| | | | Instead of potentially leaving 1 sample that requires another loop iteration.
* Update ChangeLog for mmdevapi fixChris Robinson2017-06-261-0/+2
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* Clean up some messy rounding codeChris Robinson2017-06-263-7/+30
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* Ensure the mmdevapi capture buffer is at least 100msChris Robinson2017-06-261-0/+4
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* Add a whitenoise generator to altonegenChris Robinson2017-06-251-19/+50
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* Support 32-bit float with the recording exampleChris Robinson2017-06-251-12/+35
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* Clean up some loop variablesChris Robinson2017-06-251-29/+23
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* Use the bsinc resampler for the converterChris Robinson2017-06-254-7/+18
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* Improve traces for the mmdevapi capture conversionsChris Robinson2017-06-231-10/+8
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* Use the correct destination channel offsetChris Robinson2017-06-231-2/+2
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* Don't report any output samples for no input samplesChris Robinson2017-06-231-0/+6
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* Forward Sample_ALuint to Sample_ALintChris Robinson2017-06-231-1/+1
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* Add a recording example appChris Robinson2017-06-232-0/+377
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* Stop conversion when no more source samples are availableChris Robinson2017-06-221-3/+4
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* Trace the capture converter formats for mmdevapiChris Robinson2017-06-221-0/+6
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* Workaround log2f missing on AndroidChris Robinson2017-06-211-0/+2
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* Remove an unnecessary variableChris Robinson2017-06-211-3/+2
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* Trace if dithering is enabledChris Robinson2017-06-191-3/+7
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* Update ChangeLog for the dither-depth config optionChris Robinson2017-06-181-0/+2
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* "Convert" the QSA backend to the new APIChris Robinson2017-06-185-269/+201
| | | | | | | | | | I say "convert" because it takes the lazy way and essentially just embeds the wrappers into the backend. It's done this way because I lack the means to check any changes, even syntactically. This also means the device's ExtraData field is still needed. However, this does mean all the backends are now using the new API. Code related to the old interface can now be removed.
* Make the dithering depth configurableChris Robinson2017-06-174-24/+44
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* Apply dither separately from outputChris Robinson2017-06-171-98/+68
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* Use helpers to get data from byte streamsChris Robinson2017-06-161-51/+51
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* Round the B-Format HRTF response where the multiple is definedChris Robinson2017-06-162-4/+3
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* Update ChangeLog with the PulseAudio fixChris Robinson2017-06-151-0/+3
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* Limit device buffer based on PulseAudio's tlengthChris Robinson2017-06-151-10/+9
| | | | | | Unfortunately PulseAudio has a habit of limiting tlength, and trying to calculate the device's buffer length to write regardless of tlength could result in some amount always being writable.
* Update ChangeLog with recent fixesChris Robinson2017-06-111-0/+6
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* Don't force a fade-in when resuming a paused sourceChris Robinson2017-06-091-11/+1
| | | | | | | | This needs to be handled more automatically by the mixer to work correctly. Otherwise, requiring a property update on resume can put the source into a playing state with the mixer never playing it, due to not having valid mixing parameters and the mixing parameters not getting calculated because no updates are specified by the app (and forcing an update can break deferred updates).
* Merge pull request #123 from rdb/masterkcat2017-06-091-1/+20
|\ | | | | Implement GetProcPath for FreeBSD
| * Implement GetProcPath for FreeBSDrdb2017-06-091-1/+20
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* Calculate chorus and flanger mod delays separately from feedbackChris Robinson2017-06-072-135/+118
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* Make the late lines' delay the delay average for modulationChris Robinson2017-06-071-59/+36
| | | | | | | | | Similar to the recent chorus and flanger changes, the modulation delay now swings between -n to +n, where n is less than the delay length. This brings up a slight issue with the linear interpolation, as modff doesn't produce the correct fraction value for interpolation (it's inverted, with 0 being closer to the next sample and 1 being closer to the base). So it's using nearest interpolation for now.
* Release 1.18.0openal-soft-1.18.0Chris Robinson2017-06-042-3/+3
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* Update ChangeLog about OpenSL capture supportChris Robinson2017-06-041-0/+2
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* Update default 48khz HRTF datasetChris Robinson2017-06-011-0/+0
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* Resample HRIRs prior to minimum phase reconstructionChris Robinson2017-06-011-36/+41
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* Remove unused HIDDEN_DECL macroChris Robinson2017-05-302-5/+0
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* Restrict ClampedDist to RefDistance for invalid distance attenuationChris Robinson2017-05-301-4/+10
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* Use an RMS limit of -3dB for the output limiterChris Robinson2017-05-291-2/+2
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* Add a missing includeChris Robinson2017-05-281-0/+1
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* Use peak limiting rather than RMS detectionChris Robinson2017-05-271-1/+1
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* Fix source sends' initial HF absorption and decay calculationChris Robinson2017-05-273-23/+17
| | | | | | | | The HF absorption is applied given the source distance, as relative to the source's immediate environment, with additional absorption being applied given the room/reverb environment. This does double up the amount of absorption compared to the dry path, but it can be assumed the initial reflections travel a longer distance.
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