aboutsummaryrefslogtreecommitdiffstats
path: root/Alc
Commit message (Collapse)AuthorAgeFilesLines
* Properly queue buffers for OpenSL captureChris Robinson2018-09-171-32/+9
|
* Fix some potential race conditions with OpenSLChris Robinson2018-09-171-42/+62
| | | | | | | | | For playback, increment the ring buffer's write pointer before queueing audio, to handle cases where the callback is invoked, advancing the read pointer, before the write pointer is advanced. For capture, limit the number of re-queued chunks to the number of fully read chunks.
* Handle the bsinc C resampler like the othersChris Robinson2018-09-174-48/+37
|
* Fix buffer queue mixing logicChris Robinson2018-09-161-11/+13
| | | | | | In particular, the source sample position was reduced by the size of the next buffer list item when one is completed, rather than the size of the one it just completed.
* The last reverb loop update doesn't need an aligned count.Chris Robinson2018-09-111-4/+7
|
* Ensure the max reverb update size is a multiple of 4Chris Robinson2018-09-111-2/+4
| | | | | | It's not an issue for the final mix, but if one loop has an unaligned count, the next loop will have unaligned input and output buffer targets which can crash the SSE mixers.
* Separate the delay line feeding from readingChris Robinson2018-09-101-2/+4
| | | | | Since it feeds a different line than it reads, the feeding could overwrite what's subsequently read.
* Avoid a couple line count assumptionsChris Robinson2018-09-101-2/+2
|
* Combine identical loops into a separate functionChris Robinson2018-09-101-46/+21
|
* Handle the early reflection delay separate from late refeedChris Robinson2018-09-101-57/+40
|
* Some suggested changes for iOSChris Robinson2018-09-071-3/+11
|
* PATH_MAX not MAX_PATHChris Robinson2018-09-071-1/+1
|
* Load .alsoftrc from the app bundle root on macOSChris Robinson2018-09-071-0/+23
| | | | | Not sure what priority this should have. Currently it loads after system-level configs, and before user-level configs.
* Pass the device name list to the backend probe methodChris Robinson2018-09-0719-142/+106
|
* Implement capture support for SoundIOChris Robinson2018-09-071-6/+266
|
* Rename ALCsndioBackendChris Robinson2018-09-073-51/+51
|
* Slightly restructure some loopsChris Robinson2018-09-033-39/+34
|
* Extract SIMD values right before using themChris Robinson2018-09-033-21/+21
|
* Avoid double-resizing when appending a char to a stringChris Robinson2018-09-021-2/+2
|
* Read the whole Features string for neon supportChris Robinson2018-09-021-12/+20
|
* Update a commentChris Robinson2018-08-311-1/+1
|
* Include the mid-band decay with the density gainChris Robinson2018-08-311-7/+7
|
* Calcualte and use the maximum reverb update sizeChris Robinson2018-08-311-6/+14
| | | | | Instead of requiring it to be at least as big as MAX_UPDATE_SAMPLES, which may not be true in some situations.
* Apply the decay gain when reading from the late lineChris Robinson2018-08-311-10/+17
|
* Don't modify the late reverb density with the echo parametersChris Robinson2018-08-301-13/+7
|
* Widen the plain stereo widthChris Robinson2018-08-291-2/+2
| | | | | | | | | | | Now full right and left is +-60 degrees instead of +-30. This should help create a smoother panning for a sound moving in front for plain stereo output (surround sound and HRTF are not changed). Multi-channel sources are also not affected by this change. The stereo channel defaults of +-pi/6 (30 degrees) still correspond to full left/right panning. This is an unfortuante discrepancy, but is necessary for AL_EXT_STEREO_ANGLES to work.
* Use a separate method to warp the azimuth for plain stereo outputChris Robinson2018-08-292-21/+15
|
* Use shuffle+cvt to extract SIMD values instead of storing to memoryChris Robinson2018-08-261-2/+5
|
* Avoid making extraneous callsChris Robinson2018-08-261-3/+3
|
* Pack two arrays into oneChris Robinson2018-08-261-6/+10
|
* Further limit the upper frequency range for autowahChris Robinson2018-08-261-1/+1
|
* Append the OpenSL capture device to the appropriate listChris Robinson2018-08-231-1/+1
|
* Use second-order biquad filters for the reverb's T60 decayChris Robinson2018-08-121-309/+18
|
* Store the reverb state offset locally while processingChris Robinson2018-08-111-16/+14
|
* Correctly apply reverb coefficient fading over the entire fade lengthChris Robinson2018-08-091-23/+25
|
* Crossfade the early reflections delay coefficientsChris Robinson2018-08-081-30/+50
|
* Apply late reverb density gain on late reverb inputChris Robinson2018-08-081-15/+30
| | | | | | | | | | | This also uses gain stepping for changes in density or decay, so that the late reverb tap fades smoothly between delay offsets and density gain levels simultaneously. Now with these changes, it's preferrable to apply density gain adjustments on late reverb input instead of output so that samples currently in the feedback loop won't see a sudden increase or decrease in amplitude. A similar change can probably be made for the early reflection delays to further smooth out delay changes.
* Specify both scale values to FadedDelayLineOutChris Robinson2018-08-081-9/+17
|
* Use the correct input channel for the compressor effectChris Robinson2018-08-081-1/+1
|
* Properly set early reflection all-pass coefficientChris Robinson2018-08-061-3/+7
|
* Make the all-pass coefficient part of the all-pass structureChris Robinson2018-08-061-22/+17
|
* Prevent the autowah filter frequency from reaching nyquistChris Robinson2018-08-061-1/+1
|
* Apply the vector all-pass separate on uninterleaved samplesChris Robinson2018-08-051-159/+310
| | | | | | | | | | With the vector all-pass applied in a self-contained function, the individual steps of the early and late reverb stages can be better optimized with tighter loops. That allows for more data to be held local, resulting in less thrashing from reloading the same values multiple times. There is room for further improvement, depending on the length of the early delay lines and all-pass delay lines allowing for bulk reads.
* Apply the reverb's T60 filter in groups of samplesChris Robinson2018-08-051-54/+78
| | | | | | | The late reverb line lengths are long enough to ensure a single process loop won't rely on reading samples it wrote in the same call. So we can safely read in all samples we need from the feedback buffer up front, then more efficiently filter them.
* Mix reverb samples right after generating themChris Robinson2018-08-041-31/+36
| | | | | | | Instead of generating both the early and late reverb samples first, then mixing them both to output, this now generates and mixes the early reflections then generates and mixes the late reverb. There's no reason to hold both at the same time so this reduces the amount of temporary storage needed.
* Scale the reverb decay weights to cover up to 20khzChris Robinson2018-08-031-11/+16
| | | | | | | Otherwise, using the device's maximum frequency will cause the weighting factors to shift for different sample rates, irrespective of the content being processed. 20khz is the maximum allowed reference frequency, and also acts as the upper limit of human hearing.
* Trace the NFC reference distance when setChris Robinson2018-08-031-0/+1
|
* Apply late reverb density gain adjustment on outputChris Robinson2018-08-011-22/+17
| | | | | | | | Because density/late line length changes start affecting late reverb output right away, with samples that are still going through feedback decay and not just new input samples, it makes more sense to correct for it on output instead of input. This has the additional benefit of working with the output mixer's gain fading, avoiding discontinuities from significant density gain changes.
* Fix late reverb density gain blend weightsChris Robinson2018-08-011-4/+4
| | | | | | | | | | | | Now it only accounts for the representable frequency range (0.5 normalized, or 0...pi radians instead of tau). Previously, the bulk of the weighting factors was given to the HF decay (nearly 90%, given a 44.1khz sample rate and the default 5khz reference), with low- and mid-frequency decays splitting the remaining 10%. Now it's closer to 75%, matching the range of representable frequencies above the reference. This could probably be improved further due to human hearing being less sensitive to higher frequencies, but that is much more complicated.
* Use just the omnidirectional response for the compressor effectChris Robinson2018-07-291-64/+58
| | | | | | | | This is not the output compressor/limiter, but the EFX effect. Consequently, it simply compresses the dynamic range around 1.0 (boosting samples below it by up to double, reducing samples above it by as much as half). This is not intended to prevent clipping on the output, but to instead reduce the range between quiet sounds and loud sounds.
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-05 00:22:25 +0000