aboutsummaryrefslogtreecommitdiffstats
path: root/Alc/ALc.c
Commit message (Collapse)AuthorAgeFilesLines
* 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
|
* Add a new compressor/limiterChris Robinson2017-05-271-4/+17
| | | | | | 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.
* Finalize ALC_SOFT_output_limiterChris Robinson2017-05-251-1/+1
|
* Add an option to dither 8- and 16-bit outputChris Robinson2017-05-231-0/+8
|
* Log whether the output limiter is enabled or disabledChris Robinson2017-05-111-0/+1
|
* Finalize AL_SOFT_source_spatializeChris Robinson2017-05-111-1/+1
|
* 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.
* Calculate the output limiter gain using the RMSChris Robinson2017-05-051-7/+19
|
* Start an extension to change the source's spatialize propertyChris Robinson2017-05-051-1/+4
|
* Finalize AL_SOFT_source_resamplerChris Robinson2017-05-031-1/+1
|
* Allow querying the output limiter stateChris Robinson2017-04-301-2/+12
|
* Start an extension to toggle the output limiterChris Robinson2017-04-301-5/+18
|
* Implement a limiter on the device outputChris Robinson2017-04-261-3/+11
| | | | | | This reduces the output volume when the mixed samples extend outside of -1,+1, to prevent excessive clipping. It can reduce the volume by -80dB in 50ms, and increase it by +80dB in 1s (it will not go below -80dB or above 0dB).
* Add the ability to change the source resamplerChris Robinson2017-04-211-1/+2
|
* Add a method to enumerate resamplersChris Robinson2017-04-211-0/+7
|
* Allocate a new context's voices after updating the device paramsChris Robinson2017-04-191-26/+15
|
* Use a different way to get the size of structs with flexible array membersChris Robinson2017-04-181-3/+3
|
* Store the source prop updates with the mixer voiceChris Robinson2017-04-171-21/+36
| | | | Also move its declaration and rename it for consistency.
* Trace unhandled device reset attributesChris Robinson2017-04-161-111/+109
|
* Correctly handle the attribute array size for alcGetInteger64vSOFTChris Robinson2017-04-151-2/+2
|
* Allow increasing the maximum source limitChris Robinson2017-04-141-15/+59
| | | | | | | | If the requested number of mono and stereo sources exceeds 256, the source limit will be expanded. Any config file setting overrides this. If the device is reset to have fewer sources than are currently allocated, excess sources will remain and be usable as normal, but no more can be generated until enough are delated to go back below the limit.
* Use separate atomic macros for pointersChris Robinson2017-04-141-19/+17
|
* Store the ambisonic order separate from the channel enumChris Robinson2017-04-121-92/+65
|
* Trace the capture device formatChris Robinson2017-04-101-0/+4
|
* Convert the CoreAudio backend to the updated backend APIChris Robinson2017-04-091-1/+1
|
* Reference count HRTFs and unload them when unusedChris Robinson2017-04-061-1/+10
|
* Load HRTF files as neededChris Robinson2017-04-051-32/+8
| | | | | Currently only applies to external files, rather than embedded datasets. Also, HRTFs aren't unloaded after being loaded, until library shutdown.
* Store the loaded hrtf entry container in the enumerated hrtf entryChris Robinson2017-04-051-5/+5
|
* Rename al_string_* functions to alstr_*Chris Robinson2017-04-041-19/+19
|
* Make ReleaseContext return if any contexts still remainChris Robinson2017-03-281-10/+25
|
* Use an array of pointers for effects instead of a linked listChris Robinson2017-03-271-1/+11
|
* Fix setting Ambi formats for loopback devicesChris Robinson2017-03-211-1/+1
|
* Use an atomic flag to test if a source needs to updateChris Robinson2017-03-201-1/+1
|
* Don't defer source state or offset changesChris Robinson2017-03-191-32/+3
|
* Fix alcGetInteger64vSOFT to handle ambisonic attributesChris Robinson2017-03-181-14/+28
|
* Replace a couple ALuint with ALsizeiChris Robinson2017-03-171-2/+2
|
* Remove a couple unneeded typedefsChris Robinson2017-03-141-23/+25
|
* Dynamically allocate the device's HRTF stateChris Robinson2017-03-101-37/+42
|
* Fix building without C11Chris Robinson2017-03-101-1/+2
|
* Implement NFC filters for Ambisonic renderingChris Robinson2017-03-101-0/+20
| | | | | | | | | | | | | | NFC filters currently only work when rendering to ambisonic buffers, which includes HQ rendering and ambisonic output. There are two new config options: 'decoder/nfc' (default on) enables or disables use of NFC filters globally, and 'decoder/nfc-ref-delay' (default 0) specifies the reference delay parameter for NFC-HOA rendering with ambisonic output (a value of 0 disables NFC). Currently, NFC filters rely on having an appropriate value set for AL_METERS_PER_UNIT to get the correct scaling. HQ rendering uses the averaged speaker distances as a control/reference, and currently doesn't correct for individual speaker distances (if the speakers are all equidistant, this is fine, otherwise per-speaker correction should be done as well).
* Remove an unnecessary variableChris Robinson2017-03-091-2/+1
|
* Make the voice's source pointer atomicChris Robinson2017-03-051-1/+1
|
* Increment MixCount in UpdateClockBaseChris Robinson2017-02-281-1/+4
| | | | This is to protect clocktime reads since the backend lock won't protect it.
* Dynamically allocate the channel delay buffersChris Robinson2017-02-281-0/+42
|
* Start a ALC_SOFT_loopback2 extensionChris Robinson2017-02-281-50/+190
| | | | | | | | | | | | | | | | | | | | | | This extends the base ALC_SOFT_loopback extension with support for B-Format. When ALC_FORMAT_CHANNELS_SOFT is set to ALC_BFORMAT3D_SOFT, then additional attributes must be specified. ALC_AMBISONIC_LAYOUT_SOFT must be set to ALC_ACN_SOFT or ALC_FUMA_SOFT for the desired channel layout, ALC_AMBISONIC_SCALING_SOFT must be set to ALC_N3D_SOFT, ALC_SN3D_SOFT, or ALC_FUMA_SOFT for the desired channel scaling/normalization scheme, and ALC_AMBISONIC_ORDER_SOFT must be set to an integer value greater than 0 for the ambisonic order (maximum allowed is implementation-dependent). Note that the number of channels required for ALC_BFORMAT3D_SOFT is dependent on the ambisonic order. The number of channels can be calculated by: num_channels = (order+1) * (order+1); /* or pow(order+1, 2); */ In addition, a new alcIsAmbisonicFormatSupportedSOFT function allows apps to determine which layout/scaling/order combinations are supported by the loopback device. For example, alcIsAmbisonicFormatSupported(device, ALC_ACN_SOFT, ALC_SN3D_SOFT, 2) will check if 2nd order AmbiX (ACN layout and SN3D scaling) rendering is supported for ALC_BFORMAT3D_SOFT output.
* Use a variable counter for an array size limitChris Robinson2017-02-281-21/+13
|
* Print WARNs when a device or context error is generatedChris Robinson2017-02-271-0/+1
|
* Use separate enums for the ambisonic channel order and normalizationChris Robinson2017-02-271-6/+18
|
* Move the current buffer queue entry and play position to the voiceChris Robinson2017-02-271-12/+14
| | | | | | | | | | | | | | This has a couple behavioral changes. First and biggest is that querying AL_BUFFERS_PROCESSED from a source will always return all buffers processed when in an AL_STOPPED state. Previously all buffers would be set as processed when first becoming stopped, but newly queued buffers would *not* be indicated as processed. That old behavior was not compliant with the spec, which unequivocally states "On a source in the AL_STOPPED state, all buffers are processed." Secondly, querying AL_BUFFER on an AL_STREAMING source will now always return 0. Previously it would return the current "active" buffer in the queue, but there's no basis for that in the spec.
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-01 03:41:43 +0000