aboutsummaryrefslogtreecommitdiffstats
path: root/alc
Commit message (Collapse)AuthorAgeFilesLines
* Clear the old reverb pipeline buffer when it's doneChris Robinson2022-12-171-9/+46
|
* Rework reverb fading to toggle between pipelinesChris Robinson2022-12-171-475/+360
| | | | | | | | | | | | When non-simple properties are changed, the active reverb pipeline is switched and the new parameters set on that one. The main process function will then be set to fade out input on the old pipeline, fade in input on the new pipeline, then process and mix both pipelines. After some number of samples (calculated from its decay time), the old pipeline will stop processing. This should improve the transition from a highly reverberant environment by not harshly interpolating to the new environment, as well as better handle changes to the all-pass and T60 filters.
* Use a bitset for the enabled event flagsChris Robinson2022-12-161-4/+3
|
* Avoid inlining certain exception functionsChris Robinson2022-12-162-7/+17
|
* Avoid extraneous temporaries when filling vectorsChris Robinson2022-12-152-8/+18
|
* Avoid manually specifying FFT template parametersChris Robinson2022-12-142-5/+5
|
* Implement an al::unreachable wrapper/helperChris Robinson2022-12-131-1/+1
|
* Rename a potentially confusing member functionChris Robinson2022-12-081-2/+2
|
* Avoid setting PipeWire stream properties with separate callsChris Robinson2022-12-061-12/+11
|
* Avoid using a macro to wrap standard attributesChris Robinson2022-12-067-36/+36
|
* Avoid LIKELY/UNLIKELY macrosChris Robinson2022-12-055-23/+23
|
* Avoid some uses of the LIKELY/UNLIKELY macrosChris Robinson2022-12-051-1/+1
|
* Use standard likely/unlikely attributes when availableChris Robinson2022-12-053-12/+12
|
* Use the correct variable for reading the WASAPI playback rateChris Robinson2022-12-051-2/+2
|
* Fix building with PipeWire 0.3.49 and earlierChris Robinson2022-12-021-0/+19
|
* Add missing device_clock/source_latency enums for queryingChris Robinson2022-11-291-0/+9
|
* Use complex floats for convolution reverb FFTsChris Robinson2022-11-242-24/+23
|
* Don't prematurely change ALCdevice::BufferSizeChris Robinson2022-11-221-7/+11
|
* Fix WASAPI capture handling the background COM threadChris Robinson2022-11-191-18/+30
|
* Avoid a use-after-free in the ALSA backendChris Robinson2022-11-161-12/+12
|
* Use the correct frame size for WASAPI resamplingChris Robinson2022-11-131-2/+2
|
* Change a function to a static methodChris Robinson2022-11-132-4/+5
|
* Add a config option to prevent WASAPI from using a resamplerChris Robinson2022-11-121-1/+5
|
* Add a resampler to WASAPI outputChris Robinson2022-11-121-10/+75
| | | | | | | This allows mixing at various samples rates that WASAPI doesn't otherwise support. This is mostly helpful for users that have unnecessarily high device rates (96 or 192khz), and lets the ALC_FREQUENCY attribute or frequency config option set a lower mixing rate for more efficient processing.
* Dynamically start/stop the WASAPI message threadChris Robinson2022-11-121-86/+117
|
* Try to detect the update and buffer size from PipeWireChris Robinson2022-11-081-4/+45
|
* Make sure sio_write and sio_read return valid valuesChris Robinson2022-11-071-5/+12
|
* Add preliminary support for 7.1.4 output modesChris Robinson2022-11-0314-5/+129
| | | | | | | | I don't know how different sound APIs handle 7.1.4 ("Atmos" or "Auro3D") output, but currently it simply specifies the additional channels with the height channel labels. This isn't likely how it works for a virtualized channel bed, for playing over other with-height configurations (7.1.2, 5.1.4, etc), but this should be an okay start.
* Add functions to start sources at a particular device timeChris Robinson2022-11-033-0/+13
| | | | | | | | This starts a source at a particular device clock time, rounded to the nearest sample (really, 4th sample for SIMD reasons), allowing to start a sound at a particular point in the output instead of the next update. Unlike using negative offsets, this is not affected by pitch/velocity.
* Add the ability to start a voice at a particular timeChris Robinson2022-11-031-1/+5
|
* Add missing headerChris Robinson2022-10-221-0/+1
|
* Add a warning when using the old UHJ filter optionChris Robinson2022-10-211-2/+8
|
* Rename some variables to be less ambiguousChris Robinson2022-10-211-2/+2
|
* Add separate filter options for UHJ encoding and decodingChris Robinson2022-10-212-8/+29
|
* Add an IIR filter option for UHJ encoding/decodingChris Robinson2022-10-212-3/+5
| | | | | | This uses the reversed-allpass trick to maintain linear phase. with a 256- sample look-ahead/delay to minimize distortion. This should better preserve low frequencies while maintaining a proper phase response.
* Avoid using a modulo in a loopChris Robinson2022-10-181-10/+42
|
* Remove unnecessary includes and forward declarationsChris Robinson2022-09-231-7/+0
|
* Remove an unnecessary includeChris Robinson2022-09-221-1/+0
|
* Use PulseAudio's threaded mainloop instead of a custom oneChris Robinson2022-09-221-141/+160
|
* Correctly check if a multi-channel decoder is setChris Robinson2022-09-161-1/+1
|
* Update the built-in quad and 7.1 decodersChris Robinson2022-09-161-26/+26
| | | | | Make them first- and second-order respectively. They don't have enough speakers to really be suitable for higher orders than that.
* Don't pack ambdec coefficientsChris Robinson2022-09-121-64/+22
| | | | | | | | | Tne coefficients are placed as for full 3D ACN handling. The ChanMask just indicates which have potentially useful values. This could be a bit more agressive and clear ChanMask bits for channels that don't contribute to output, so that a decoder the specifies height-related channel bits, but leaves their coefficients all 0, can be handled as 2D. I don't expect many ambdec files to be like that, though.
* Ensure coeff_scale is set for ambdec filesChris Robinson2022-09-111-0/+1
|
* Track if doing 2D mixing onlyChris Robinson2022-09-044-6/+15
| | | | And use it to select the proper HF scales
* Allow different HF scales for 2D mixing/outputChris Robinson2022-09-042-2/+2
| | | | | Not actually used yet, the device needs to track whether it's using 2D or 3D mixing.
* Use the difference in HF scale for upsampling ambisonicsChris Robinson2022-09-032-2/+2
|
* Revert "Don't apply the HF scaling for "upsampling" ambisonics"Chris Robinson2022-09-022-4/+42
| | | | This reverts commit bf3f63fb4c5faa45784d7433d68b7013e29ee2c1.
* Use an icosahedron for 3D second-orderChris Robinson2022-09-011-32/+33
|
* Don't apply the HF scaling for "upsampling" ambisonicsChris Robinson2022-08-312-42/+4
| | | | | | | | | | | | Not sure if this is best, but it fixes the issue of upsampling lower orders to higher orders multiple times (mixing lower order to higher order, cutting back to lower order by dropping the extra channels, then upsampling again, applying the HF scale multiple times and messing up the high frequencies/energy vector), as well as having too-strong high frequencies of upsampled content in general. As it is, with the new upsampling method, the lower order signal contributes to some higher order channels, which helps offset the high frequency scaling difference (except first- to second-order, which adds nothing).
* Use a more precise 3D7.1 decoderChris Robinson2022-08-301-12/+12
|
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-13 18:32:28 +0000