aboutsummaryrefslogtreecommitdiffstats
path: root/Alc
Commit message (Collapse)AuthorAgeFilesLines
* Fix default effect initializationChris Robinson2017-07-191-28/+11
|
* Add an all-pass filter that replicates the band splitter's phase shiftChris Robinson2017-07-192-0/+51
|
* Scale the source volume by +3dB for a full spreadChris Robinson2017-07-181-6/+7
| | | | | This effectively turns a full spread source into an ambient response, preventing such sources from being unexpectedly quiet.
* Apply the output buffer offset before writing to itChris Robinson2017-07-151-24/+24
|
* Set the float PCM GUID for wave files only when outputting floatChris Robinson2017-07-151-2/+3
|
* Add the default auxiliary slot to the active slot arrayChris Robinson2017-07-132-18/+11
|
* Initialize the default effect after device updateChris Robinson2017-07-131-35/+31
|
* Use macros to set and restore the mixer FPU modeChris Robinson2017-07-133-17/+10
|
* Store the default effect slot in the contextChris Robinson2017-07-132-59/+82
|
* Store the QSA backend's ExtraData in the wrapper structChris Robinson2017-06-291-118/+121
|
* Use sqrtf for single-precision square rootsChris Robinson2017-06-291-1/+1
|
* Add casts to silence MSVCChris Robinson2017-06-291-1/+1
|
* Define a backup log2f if the compiler doesn't have itChris Robinson2017-06-291-3/+0
|
* Calculate the converter stepping value using floating pointChris Robinson2017-06-281-3/+3
|
* Remove the fastf2u conversion functionChris Robinson2017-06-276-13/+12
|
* Use a macro to apply NFC filtered mixes instead of a loopChris Robinson2017-06-261-18/+16
|
* Round the converter's stepping valueChris Robinson2017-06-261-2/+3
|
* Trace the message name in the message handler loopChris Robinson2017-06-261-1/+14
|
* Convert all input samples in the loopChris Robinson2017-06-261-2/+2
| | | | Instead of potentially leaving 1 sample that requires another loop iteration.
* Clean up some messy rounding codeChris Robinson2017-06-262-7/+12
|
* Ensure the mmdevapi capture buffer is at least 100msChris Robinson2017-06-261-0/+4
|
* Clean up some loop variablesChris Robinson2017-06-251-29/+23
|
* Use the bsinc resampler for the converterChris Robinson2017-06-253-7/+16
|
* Improve traces for the mmdevapi capture conversionsChris Robinson2017-06-231-10/+8
|
* Use the correct destination channel offsetChris Robinson2017-06-231-2/+2
|
* Don't report any output samples for no input samplesChris Robinson2017-06-231-0/+6
|
* Forward Sample_ALuint to Sample_ALintChris Robinson2017-06-231-1/+1
|
* Stop conversion when no more source samples are availableChris Robinson2017-06-221-3/+4
|
* Trace the capture converter formats for mmdevapiChris Robinson2017-06-221-0/+6
|
* Workaround log2f missing on AndroidChris Robinson2017-06-211-0/+2
|
* Remove an unnecessary variableChris Robinson2017-06-211-3/+2
|
* Trace if dithering is enabledChris Robinson2017-06-191-3/+7
|
* "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
|
* Apply dither separately from outputChris Robinson2017-06-171-98/+68
|
* Use helpers to get data from byte streamsChris Robinson2017-06-161-51/+51
|
* Round the B-Format HRTF response where the multiple is definedChris Robinson2017-06-162-4/+3
|
* 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
|
* Calculate chorus and flanger mod delays separately from feedbackChris Robinson2017-06-072-135/+118
|
* 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
|
* Use an RMS limit of -3dB for the output limiterChris Robinson2017-05-291-2/+2
|
* Use peak limiting rather than RMS detectionChris Robinson2017-05-271-1/+1
|
* 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
|