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* Trace if dithering is enabledChris Robinson2017-06-191-3/+7
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* "Convert" the QSA backend to the new APIChris Robinson2017-06-184-250/+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 Robinson2017-06-172-20/+33
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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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* 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.
* 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.
* 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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* 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 Robinson2017-05-271-21/+15
| | | | | | | | 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.
* Use normal air absorption for the sendsChris Robinson2017-05-271-1/+1
| | | | | Applies just for the normal air absorption which uses the air absorption factor, not the automated decay applied when WetGainAuto is enabled.
* Add a new compressor/limiterChris Robinson2017-05-273-97/+274
| | | | | | This is just for the output limiter right now, but in the future can be used for the compressor EFX effect. The parameters are also hardcoded, but can be made configurable after 1.18.
* Fix handling chorus and flanger LFO displacement offsetChris Robinson2017-05-262-2/+8
| | | | | The phase offset is modulo-wrapped rather than masked, so it's best to avoid negative offsets.
* Properly handle the chorus and flanger LFOsChris Robinson2017-05-262-24/+30
| | | | | The effects' specified delay is the average delay time, meaning the delay offset should move between -n and +n relative to the delay, where n <= delay.
* Finalize ALC_SOFT_output_limiterChris Robinson2017-05-251-1/+1
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* Apply distance compensation separatelyChris Robinson2017-05-251-43/+66
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* Add a config option to specify custom ALSA devicesChris Robinson2017-05-231-5/+40
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* Add an option to dither 8- and 16-bit outputChris Robinson2017-05-232-45/+146
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* Reduce the amount of variables that hold the same valueChris Robinson2017-05-212-10/+8
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* Avoid calculating the filter coefficients multiple timesChris Robinson2017-05-211-24/+36
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* Add a method to copy a filter's coefficientsChris Robinson2017-05-212-39/+6
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* Avoid unnecessary doublesChris Robinson2017-05-211-2/+2
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* Use a macro to specify the decay target gainChris Robinson2017-05-212-7/+7
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* Use more correct doppler shift calculationsChris Robinson2017-05-201-11/+24
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* Restore spec-defined cone behavior for auxiliary sendsChris Robinson2017-05-201-26/+23
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* Apply more proper air absorption to the wet pathChris Robinson2017-05-191-21/+53
| | | | | | | | This properly accounts for the room rolloff factor for normal air absorption (which makes it none by default, like distance attenuation), and uses the reverb's decay time, decay hf ratio, decay hf limit, and room air absorption properties to calculate an initial hf decay with the WetGainAuto flag. This mirrors the behavior of the initial distance decay.
* Reorganize some code to have fewer temporariesChris Robinson2017-05-171-39/+34
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* Add an env var to specify a default pulse deviceChris Robinson2017-05-161-0/+7
| | | | | Some apps don't allow selecting an audio device, and due to problems with KDE, PulseAudio isn't allowed to move the stream after being created by default.
* Reduce the main reverb filter gain limit to match the restChris Robinson2017-05-161-4/+4
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* Limit the dry and wet path filter gains to -60dBChris Robinson2017-05-151-4/+4
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* Improve distance-related absorption and decay attenuationChris Robinson2017-05-151-15/+14
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* Make reverb delay line structs use interleaved floatsChris Robinson2017-05-151-68/+73
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* Make the reverb's early and late feedback lines interleavedChris Robinson2017-05-141-42/+47
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* Don't enable NFC for a 0 reference delayChris Robinson2017-05-141-1/+1
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* Update a couple comments about the reverb modulationChris Robinson2017-05-131-12/+13
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* Replace 4 separate all-passes with one vector all-passChris Robinson2017-05-131-53/+47
| | | | | | Each 4 related all-passes now share a structure with one delay line, which uses an interleaved sample history. Also fixes some potential rounding problems for delay lines with interleaved samples.
* Apply reverb modulation to the late feedback linesChris Robinson2017-05-121-104/+71
| | | | | | | | This seems to be more in-line with the intended behavior, to allow build-up and overlap within the reverb decay, rather than a pitch-shift on input. Unfortunately there's no readily available implementation of this reverb model that includes modulation to compare with, so a low depth coefficient is used to keep it very subtle.
* Remove the 0.5 gain on the reverb outputChris Robinson2017-05-111-6/+2
| | | | | Turns out to be unnecessary, as it reduced the volume below what other reverb implementations provide with the same presets.
* Restore the previous reverb B2A and A2B matricesChris Robinson2017-05-111-12/+12
| | | | Also, untranspose the A2B matrix.
* Log whether the output limiter is enabled or disabledChris Robinson2017-05-111-0/+1
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* Finalize AL_SOFT_source_spatializeChris Robinson2017-05-111-1/+1
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* Update ALC_OUTPUT_LIMITER_SOFT to handle ALC_DONT_CARE_SOFTChris Robinson2017-05-111-4/+8
| | | | | Essentially just adding a comment about it. Since we default to on, the behavior already fit.
* Reduce LIMITER_VALUE_MAXChris Robinson2017-05-091-6/+12
| | | | | | The previous value couldn't actually be expressed as a float and got rounded up to the next whole number value, leaving the potential for an overrun in the squared sum.
* Store the output limiter values as fixed-point integersChris Robinson2017-05-081-11/+6
| | | | | This helps keep the squared sum stable over larger updates, also avoiding the need to keep recalculating it.
* Update reverb conversion matrices and output gainChris Robinson2017-05-081-10/+12
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* Apply attenuation when downmixing multi-channel sounds for panningChris Robinson2017-05-071-10/+26
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