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|Name: libopenal1||Distribution: openSUSE Tumbleweed|
|Version: 1.22.2||Vendor: openSUSE|
|Release: 1.2||Build date: Thu Sep 1 02:53:57 2022|
|Group: System/Libraries||Build host: lamb14|
|Size: 980759||Source RPM: openal-soft-1.22.2-1.2.src.rpm|
|Summary: Audio library with an OpenGL-resembling API|
OpenAL is an audio library designed in the spirit of the OpenGL API. OpenAL provides capabilities for playing audio in a virtual 3D environment. Distance attenuation, doppler shift, and directional sound emitters are among the features handled by the API. More advanced effects, including air absorption, occlusion, and environmental reverb, are available through the EFX extension. It also facilitates streaming audio, multi-channel buffers, and audio capture.
* Tue Aug 16 2022 Dirk Müller <firstname.lastname@example.org> - disable pipewire backend to avoid buildcycle ffmpeg-4, libopenmpt, mpg123, openal-soft, pipewire * Mon Aug 01 2022 Dirk Müller <email@example.com> - update to 1.22.2: * Fixed PipeWire version check. * Fixed building with PipeWire versions before 0.3.33. * Fixed CoreAudio capture. * Fixed air absorption strength. * Fixed ALSA not being used on some systems without PipeWire and PulseAudio. * Fixed OpenSL capturing noise. * Fixed Oboe capture failing with some buffer sizes. * Added checks for the runtime PipeWire version. * The same or newer version than is used for building will be needed at runtime for the backend to work. * Separated 3D7.1 into its own speaker configuration. * Implemented the ALC_SOFT_reopen_device extension. * This allows for moving devices to different outputs without losing object state. * Implemented the ALC_SOFT_output_mode extension. * Implemented the AL_SOFT_callback_buffer extension. * Implemented the AL_SOFT_UHJ extension. * This supports native UHJ buffer formats and Super Stereo processing. * Implemented the legacy EAX extensions. * Enabled by default only on Windows. * Improved sound positioning stability when a source is near the listener. * Improved the default 5.1 output decoder. * Improved the high frequency response for the HRTF second-order ambisonic decoder. * Improved SoundIO capture behavior. * Fixed UHJ output on NEON-capable CPUs. * Fixed redundant effect updates when setting an effect property to the current value. * Fixed WASAPI capture using really low sample rates, and sources with very high pitch shifts when using a bsinc resampler. * Added a PipeWire backend. * Added enumeration for the JACK and CoreAudio backends. * Added optional support for RTKit to get real-time priority. * Added an option for JACK playback to render directly in the real-time processing callback. * Added an option for custom JACK devices. * Added utilities to encode and decode UHJ audio files. * Added an in-progress extension to hold sources in a playing state when a device disconnects. * Lowered the priority of the JACK backend. - drop openal-soft-gcc11.diff (obsolete) * Mon Jul 04 2022 Jan Engelhardt <firstname.lastname@example.org> - Remove -msse2 from the i586 gcc command lines. * Wed Feb 17 2021 Ludwig Nussel <email@example.com> - fix gcc11 build (openal-soft-gcc11.diff) * Tue Feb 09 2021 Dirk Müller <firstname.lastname@example.org> - update to 1.21.1: * Improved alext.h's detection of standard types. * Improved slightly the local source position when the listener and source are near each other. * Improved click/pop prevention for sounds that stop prematurely. * Fixed compilation for Windows ARM targets with MSVC. * Fixed ARM NEON detection on Windows. * Fixed CoreAudio capture when the requested sample rate doesn't match the system configuration. * Fixed OpenSL capture desyncing from the internal capture buffer. * Fixed sources missing a batch update when applied after quickly restarting the source. * Fixed missing source stop events when stopping a paused source. * Added capture support to the experimental Oboe backend. * Sat Jan 16 2021 Matthias Mailänder <email@example.com> - new version 1.21.0 * Updated library codebase to C++14. * Implemented the AL_SOFT_effect_target extension. * Implemented the AL_SOFT_events extension. * Implemented the ALC_SOFT_loopback_bformat extension. * Improved memory use for mixing voices. * Improved detection of NEON capabilities. * Improved handling of PulseAudio devices that lack manual start control. * Improved mixing performance with PulseAudio. * Improved high-frequency scaling quality for the HRTF B-Format decoder. * Improved makemhr's HRIR delay calculation. * Improved WASAPI capture of mono formats with multichannel input. * Reimplemented the modulation stage for reverb. * Enabled real-time mixing priority by default, for backends that use the setting. It can still be disabled in the config file. * Enabled dual-band processing for the built-in quad and 7.1 output decoders. * Fixed a potential crash when deleting an effect slot immediately after the last source using it stops. * Fixed building with the static runtime on MSVC. * Fixed using source stereo angles outside of -pi...+pi. * Fixed the buffer processed event count for sources that start with empty buffers. * Fixed trying to open an unopenable WASAPI device causing all devices to stop working. * Fixed stale devices when re-enumerating WASAPI devices. * Fixed using unicode paths with the log file on Windows. * Fixed DirectSound capture reporting bad sample counts or erroring when reading samples. * Added an in-progress extension for a callback-driven buffer type. * Added an in-progress extension for higher-order B-Format buffers. * Added an in-progress extension for convolution reverb. * Added an experimental Oboe backend for Android playback. This requires the Oboe sources at build time, so that it's built as a static library included in libopenal. * Added an option for auto-connecting JACK ports. * Added greater-than-stereo support to the SoundIO backend. * Modified the mixer to be fully asynchronous with the external API, and should now be real-time safe. Although alcRenderSamplesSOFT is not due to locking to check the device handle validity. * Modified the UHJ encoder to use an all-pass FIR filter that's less harmful to non-filtered signal phase. * Converted examples from SDL_sound to libsndfile. To avoid issues when combining SDL2 and SDL_sound. * Worked around a 32-bit GCC/MinGW bug with TLS destructors. See: https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562 * Reduced the maximum number of source sends from 16 to 6. * Removed the QSA backend. It's been broken for who knows how long. * Got rid of the compile-time native-tools targets, using cmake and global initialization instead. This should make cross-compiling less troublesome. * Sat Jul 04 2020 Matthias Mailänder <firstname.lastname@example.org> - Add SDL2 and PortAudio backends * Tue Feb 04 2020 Ludwig Nussel <email@example.com> - new version 1.20.1 The changes from 1.20.0 include: * Implemented the AL_SOFT_direct_channels_remix extension. * This extends AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have a matching output channel. * Implemented the AL_SOFT_bformat_ex extension. * This extends B-Format buffer support for N3D or SN3D scaling, or ACN channel ordering. * Fixed a potential voice leak when a source is started and stopped or restarted in quick succession. * Fixed a potential device reset failure with JACK. * Improved handling of unsupported channel configurations with WASAPI. * Such setups will now try to output at least a stereo mix. * Improved clarity a bit for the HRTF second-order ambisonic decoder. * Improved detection of compatible layouts for SOFA files in makemhr and sofa-info. * Added the ability to resample HRTFs on load. * MHR files no longer need to match the device sample rate to be usable. * Added an option to limit the HRTF's filter length. The changes from 1.19.1 include: * Converted the library codebase to C++11. * A lot of hacks and custom structures have been replaced with standard or cleaner implementations. * Partially implemented the Vocal Morpher effect. * Fixed the bsinc SSE resamplers on non-GCC compilers. * Fixed OpenSL capture. * Fixed support for extended capture formats with OpenSL. * Fixed handling of WASAPI not reporting a default device. * Fixed performance problems relating to semaphores on macOS. * Modified the bsinc12 resampler's transition band to better avoid aliasing noise. * Modified alcResetDeviceSOFT to attempt recovery of disconnected devices. * Modified the virtual speaker layout for HRTF B-Format decoding. * Modified the PulseAudio backend to use a custom processing loop. * Renamed the makehrtf utility to makemhr. * Improved the efficiency of the bsinc resamplers when up-sampling. * Improved the quality of the bsinc resamplers slightly. * Improved the efficiency of the HRTF filters. * Improved the HRTF B-Format decoder coefficient generation. * Improved reverb feedback fading to be more consistent with pan fading. * Improved handling of sources that end prematurely, avoiding loud clicks. * Improved the performance of some reverb processing loops. * Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of some quality. * Notably, down-sampling has less smooth pitch ramping. * Added support for SOFA input files with makemhr. * Added a build option to use pre-built native tools. * For cross-compiling, use with caution and ensure the native tools' binaries are kept up-to-date. * Added an adjust-latency config option for the PulseAudio backend. * Added basic support for multi-field HRTFs. * Added an option for mixing first- or second-order B-Format with HRTF output. * This can improve HRTF performance given a number of sources. * Added an RC file for proper DLL version information. * Disabled some old KDE workarounds by default. * Specifically, PulseAudio streams can now be moved (KDE may try to move them after opening). - makehrtf tool was renamed to makemhr - disable jack backend as it doesn't work due to missing jack_error_callback * Wed May 29 2019 Martin Pluskal <firstname.lastname@example.org> - Use more of macros for building - Build gui config tool as well * Sat Apr 06 2019 Jan Engelhardt <email@example.com> - Trim bias from description, trim metadata duplication from description, trim main description repetition in lesser subpackages' description. Spruce up summaries. Fix SRPM group. - Add missing Requires inside baselibs.conf. - Remove insatisfiable Recommends. Add Provides/Conflicts for the move of makehrtf. * Thu Mar 21 2019 Stefan Brüns <firstname.lastname@example.org> - Packaging changes: * Move makehrtf from the devel package to a separate package, as it is the only part not under LGPL (or MIT). * Move the remaining tools and data files to separate packages, to get the License tag correct, and make the data files noarch. * Use https in Url and Source tags. - Update to 1.19.1 * The changes from 1.19.0 include: - Implemented capture support for the SoundIO backend. - Fixed source buffer queues potentially not playing properly when a queue entry completes. - Fixed possible unexpected failures when generating auxiliary effect slots. - Fixed a crash with certain reverb or device settings. - Fixed OpenSL capture. - Improved output limiter response, better ensuring the sample amplitude is clamped for output. * The changes from 1.18.2 include: - Implemented the ALC_SOFT_device_clock extension. - Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects. - Fixed compiling on FreeBSD systems that use freebsd-lib 9.1. - Fixed compiling on NetBSD. - Fixed the reverb effect's density scale and panning parameters. - Fixed use of the WASAPI backend with certain games, which caused odd COM initialization errors. - Increased the number of virtual channels for decoding Ambisonics to HRTF output. - Changed 32-bit x86 builds to use SSE2 math by default for performance. Build-time options are available to use just SSE1 or x87 instead. - Replaced the 4-point Sinc resampler with a more efficient cubic resampler. - Renamed the MMDevAPI backend to WASAPI. - Added support for 24-bit, dual-ear HRTF data sets. The built-in data set has been updated to 24-bit. - Added a 24- to 48-point band-limited Sinc resampler. - Added an SDL2 playback backend. Disabled by default to avoid a dependency on SDL2. - Improved the performance and quality of the Chorus and Flanger effects. - Improved the efficiency of the band-limited Sinc resampler. - Improved the Sinc resampler's transition band to avoid over-attenuating higher frequencies. - Improved the performance of some filter operations. - Improved the efficiency of object ID lookups. - Improved the efficienty of internal voice/source synchronization. - Improved AL call error logging with contextualized messages. - Removed the reverb effect's modulation stage. Due to the lack of reference for its intended behavior and strength.
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