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654 lines
21 KiB
654 lines
21 KiB
openal-soft-1.22.2:
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Fixed PipeWire version check.
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Fixed building with PipeWire versions before 0.3.33.
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openal-soft-1.22.1:
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Fixed CoreAudio capture.
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Fixed air absorption strength.
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Fixed handling 5.1 devices on Windows that use Rear channels instead of
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Side channels.
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Fixed some compilation issues on MinGW.
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Fixed ALSA not being used on some systems without PipeWire and PulseAudio.
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Fixed OpenSL capturing noise.
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Fixed Oboe capture failing with some buffer sizes.
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Added checks for the runtime PipeWire version. The same or newer version
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than is used for building will be needed at runtime for the backend to
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work.
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Separated 3D7.1 into its own speaker configuration.
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openal-soft-1.22.0:
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Implemented the ALC_SOFT_reopen_device extension. This allows for moving
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devices to different outputs without losing object state.
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Implemented the ALC_SOFT_output_mode extension.
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Implemented the AL_SOFT_callback_buffer extension.
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Implemented the AL_SOFT_UHJ extension. This supports native UHJ buffer
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formats and Super Stereo processing.
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Implemented the legacy EAX extensions. Enabled by default only on Windows.
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Improved sound positioning stability when a source is near the listener.
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Improved the default 5.1 output decoder.
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Improved the high frequency response for the HRTF second-order ambisonic
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decoder.
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Improved SoundIO capture behavior.
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Fixed UHJ output on NEON-capable CPUs.
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Fixed redundant effect updates when setting an effect property to the
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current value.
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Fixed WASAPI capture using really low sample rates, and sources with very
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high pitch shifts when using a bsinc resampler.
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Added a PipeWire backend.
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Added enumeration for the JACK and CoreAudio backends.
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Added optional support for RTKit to get real-time priority. Only used as a
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backup when pthread_setschedparam fails.
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Added an option for JACK playback to render directly in the real-time
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processing callback. For lower playback latency, on by default.
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Added an option for custom JACK devices.
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Added utilities to encode and decode UHJ audio files. Files are decoded to
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the .amb format, and are encoded from libsndfile-compatible formats.
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Added an in-progress extension to hold sources in a playing state when a
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device disconnects. Allows devices to be reset or reopened and have sources
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resume from where they left off.
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Lowered the priority of the JACK backend. To avoid it getting picked when
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PipeWire is providing JACK compatibility, since the JACK backend is less
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robust with auto-configuration.
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openal-soft-1.21.1:
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Improved alext.h's detection of standard types.
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Improved slightly the local source position when the listener and source
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are near each other.
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Improved click/pop prevention for sounds that stop prematurely.
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Fixed compilation for Windows ARM targets with MSVC.
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Fixed ARM NEON detection on Windows.
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Fixed CoreAudio capture when the requested sample rate doesn't match the
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system configuration.
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Fixed OpenSL capture desyncing from the internal capture buffer.
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Fixed sources missing a batch update when applied after quickly restarting
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the source.
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Fixed missing source stop events when stopping a paused source.
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Added capture support to the experimental Oboe backend.
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openal-soft-1.21.0:
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Updated library codebase to C++14.
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Implemented the AL_SOFT_effect_target extension.
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Implemented the AL_SOFT_events extension.
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Implemented the ALC_SOFT_loopback_bformat extension.
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Improved memory use for mixing voices.
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Improved detection of NEON capabilities.
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Improved handling of PulseAudio devices that lack manual start control.
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Improved mixing performance with PulseAudio.
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Improved high-frequency scaling quality for the HRTF B-Format decoder.
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Improved makemhr's HRIR delay calculation.
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Improved WASAPI capture of mono formats with multichannel input.
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Reimplemented the modulation stage for reverb.
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Enabled real-time mixing priority by default, for backends that use the
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setting. It can still be disabled in the config file.
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Enabled dual-band processing for the built-in quad and 7.1 output decoders.
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Fixed a potential crash when deleting an effect slot immediately after the
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last source using it stops.
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Fixed building with the static runtime on MSVC.
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Fixed using source stereo angles outside of -pi...+pi.
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Fixed the buffer processed event count for sources that start with empty
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buffers.
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Fixed trying to open an unopenable WASAPI device causing all devices to
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stop working.
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Fixed stale devices when re-enumerating WASAPI devices.
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Fixed using unicode paths with the log file on Windows.
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Fixed DirectSound capture reporting bad sample counts or erroring when
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reading samples.
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Added an in-progress extension for a callback-driven buffer type.
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Added an in-progress extension for higher-order B-Format buffers.
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Added an in-progress extension for convolution reverb.
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Added an experimental Oboe backend for Android playback. This requires the
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Oboe sources at build time, so that it's built as a static library included
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in libopenal.
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Added an option for auto-connecting JACK ports.
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Added greater-than-stereo support to the SoundIO backend.
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Modified the mixer to be fully asynchronous with the external API, and
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should now be real-time safe. Although alcRenderSamplesSOFT is not due to
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locking to check the device handle validity.
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Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
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to non-filtered signal phase.
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Converted examples from SDL_sound to libsndfile. To avoid issues when
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combining SDL2 and SDL_sound.
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Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
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https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
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Reduced the maximum number of source sends from 16 to 6.
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Removed the QSA backend. It's been broken for who knows how long.
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Got rid of the compile-time native-tools targets, using cmake and global
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initialization instead. This should make cross-compiling less troublesome.
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openal-soft-1.20.1:
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Implemented the AL_SOFT_direct_channels_remix extension. This extends
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AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
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a matching output channel.
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Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
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support for N3D or SN3D scaling, or ACN channel ordering.
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Fixed a potential voice leak when a source is started and stopped or
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restarted in quick succession.
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Fixed a potential device reset failure with JACK.
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Improved handling of unsupported channel configurations with WASAPI. Such
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setups will now try to output at least a stereo mix.
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Improved clarity a bit for the HRTF second-order ambisonic decoder.
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Improved detection of compatible layouts for SOFA files in makemhr and
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sofa-info.
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Added the ability to resample HRTFs on load. MHR files no longer need to
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match the device sample rate to be usable.
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Added an option to limit the HRTF's filter length.
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openal-soft-1.20.0:
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Converted the library codebase to C++11. A lot of hacks and custom
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structures have been replaced with standard or cleaner implementations.
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Partially implemented the Vocal Morpher effect.
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Fixed the bsinc SSE resamplers on non-GCC compilers.
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Fixed OpenSL capture.
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Fixed support for extended capture formats with OpenSL.
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Fixed handling of WASAPI not reporting a default device.
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Fixed performance problems relating to semaphores on macOS.
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Modified the bsinc12 resampler's transition band to better avoid aliasing
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noise.
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Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
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Modified the virtual speaker layout for HRTF B-Format decoding.
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Modified the PulseAudio backend to use a custom processing loop.
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Renamed the makehrtf utility to makemhr.
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Improved the efficiency of the bsinc resamplers when up-sampling.
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Improved the quality of the bsinc resamplers slightly.
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Improved the efficiency of the HRTF filters.
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Improved the HRTF B-Format decoder coefficient generation.
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Improved reverb feedback fading to be more consistent with pan fading.
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Improved handling of sources that end prematurely, avoiding loud clicks.
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Improved the performance of some reverb processing loops.
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Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
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some quality. Notably, down-sampling has less smooth pitch ramping.
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Added support for SOFA input files with makemhr.
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Added a build option to use pre-built native tools. For cross-compiling,
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use with caution and ensure the native tools' binaries are kept up-to-date.
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Added an adjust-latency config option for the PulseAudio backend.
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Added basic support for multi-field HRTFs.
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Added an option for mixing first- or second-order B-Format with HRTF
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output. This can improve HRTF performance given a number of sources.
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Added an RC file for proper DLL version information.
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Disabled some old KDE workarounds by default. Specifically, PulseAudio
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streams can now be moved (KDE may try to move them after opening).
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openal-soft-1.19.1:
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Implemented capture support for the SoundIO backend.
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Fixed source buffer queues potentially not playing properly when a queue
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entry completes.
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Fixed possible unexpected failures when generating auxiliary effect slots.
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Fixed a crash with certain reverb or device settings.
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Fixed OpenSL capture.
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Improved output limiter response, better ensuring the sample amplitude is
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clamped for output.
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openal-soft-1.19.0:
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Implemented the ALC_SOFT_device_clock extension.
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Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
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Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
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Fixed compiling on NetBSD.
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Fixed the reverb effect's density scale and panning parameters.
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Fixed use of the WASAPI backend with certain games, which caused odd COM
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initialization errors.
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Increased the number of virtual channels for decoding Ambisonics to HRTF
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output.
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Changed 32-bit x86 builds to use SSE2 math by default for performance.
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Build-time options are available to use just SSE1 or x87 instead.
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Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
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Renamed the MMDevAPI backend to WASAPI.
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Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
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has been updated to 24-bit.
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Added a 24- to 48-point band-limited Sinc resampler.
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Added an SDL2 playback backend. Disabled by default to avoid a dependency
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on SDL2.
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Improved the performance and quality of the Chorus and Flanger effects.
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Improved the efficiency of the band-limited Sinc resampler.
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Improved the Sinc resampler's transition band to avoid over-attenuating
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higher frequencies.
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Improved the performance of some filter operations.
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Improved the efficiency of object ID lookups.
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Improved the efficienty of internal voice/source synchronization.
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Improved AL call error logging with contextualized messages.
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Removed the reverb effect's modulation stage. Due to the lack of reference
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for its intended behavior and strength.
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openal-soft-1.18.2:
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Fixed resetting the FPU rounding mode after certain function calls on
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Windows.
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Fixed use of SSE intrinsics when building with Clang on Windows.
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Fixed a crash with the JACK backend when using JACK1.
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Fixed use of pthread_setnane_np on NetBSD.
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Fixed building on FreeBSD with an older freebsd-lib.
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OSS now links with libossaudio if found at build time (for NetBSD).
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openal-soft-1.18.1:
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Fixed an issue where resuming a source might not restart playing it.
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Fixed PulseAudio playback when the configured stream length is much less
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than the requested length.
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Fixed MMDevAPI capture with sample rates not matching the backing device.
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Fixed int32 output for the Wave Writer.
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Fixed enumeration of OSS devices that are missing device files.
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Added correct retrieval of the executable's path on FreeBSD.
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Added a config option to specify the dithering depth.
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Added a 5.1 decoder preset that excludes front-center output.
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openal-soft-1.18.0:
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Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
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Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
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AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
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Implemented 3D processing for some effects. Currently implemented for
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Reverb, Compressor, Equalizer, and Ring Modulator.
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Implemented 2-channel UHJ output encoding. This needs to be enabled with a
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config option to be used.
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Implemented dual-band processing for high-quality ambisonic decoding.
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Implemented distance-compensation for surround sound output.
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Implemented near-field emulation and compensation with ambisonic rendering.
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Currently only applies when using the high-quality ambisonic decoder or
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ambisonic output, with appropriate config options.
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Implemented an output limiter to reduce the amount of distortion from
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clipping.
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Implemented dithering for 8-bit and 16-bit output.
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Implemented a config option to select a preferred HRTF.
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Implemented a run-time check for NEON extensions using /proc/cpuinfo.
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Implemented experimental capture support for the OpenSL backend.
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Fixed building on compilers with NEON support but don't default to having
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NEON enabled.
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Fixed support for JACK on Windows.
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Fixed starting a source while alcSuspendContext is in effect.
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Fixed detection of headsets as headphones, with MMDevAPI.
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Added support for AmbDec config files, for custom ambisonic decoder
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configurations. Version 3 files only.
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Added backend-specific options to alsoft-config.
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Added first-, second-, and third-order ambisonic output formats. Currently
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only works with backends that don't rely on channel labels, like JACK,
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ALSA, and OSS.
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Added a build option to embed the default HRTFs into the lib.
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Added AmbDec presets to enable high-quality ambisonic decoding.
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Added an AmbDec preset for 3D7.1 speaker setups.
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Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
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the provided ambdec presets.
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Added the ability for MMDevAPI to open devices given a Device ID or GUID
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string.
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Added an option to the example apps to open a specific device.
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Increased the maximum auxiliary send limit to 16 (up from 4). Requires
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requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
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attribute.
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Increased the default auxiliary effect slot count to 64 (up from 4).
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Reduced the default period count to 3 (down from 4).
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Slightly improved automatic naming for enumerated HRTFs.
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Improved B-Format decoding with HRTF output.
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Improved internal property handling for better batching behavior.
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Improved performance of certain filter uses.
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Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
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extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
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openal-soft-1.17.2:
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Implemented device enumeration for OSSv4.
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Fixed building on OSX.
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Fixed building on non-Windows systems without POSIX-2008.
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Fixed Dedicated Dialog and Dedicated LFE effect output.
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Added a build option to override the share install dir.
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Added a build option to static-link libgcc for MinGW.
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openal-soft-1.17.1:
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Fixed building with JACK and without PulseAudio.
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Fixed building on FreeBSD.
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Fixed the ALSA backend's allow-resampler option.
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Fixed handling of inexact ALSA period counts.
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Altered device naming scheme on Windows backends to better match other
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drivers.
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Updated the CoreAudio backend to use the AudioComponent API. This clears up
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deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
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openal-soft-1.17.0:
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Implemented a JACK playback backend.
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Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
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Implemented the ALC_SOFT_HRTF extension.
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Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
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Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
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24-point Sinc resampling, and performs anti-aliasing.
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Implemented B-Format output support for the wave file writer. This creates
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FuMa-style first-order Ambisonics wave files (AMB format).
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Implemented a stereo-mode config option for treating stereo modes as either
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speakers or headphones.
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Implemented per-device configuration options.
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Fixed handling of PulseAudio and MMDevAPI devices that have identical
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descriptions.
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Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
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Fixed logging of Unicode characters on Windows.
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Fixed 5.1 surround sound channels. By default it will now use the side
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channels for the surround output. A configuration using rear channels is
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still available.
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Fixed the QSA backend potentially altering the capture format.
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Fixed detecting MMDevAPI's default device.
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Fixed returning the default capture device name.
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Fixed mixing property calculations when deferring context updates.
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Altered the behavior of alcSuspendContext and alcProcessContext to better
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match certain Windows drivers.
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Altered the panning algorithm, utilizing Ambisonics for better side and
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back positioning cues with surround sound output.
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Improved support for certain older Windows apps.
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Improved the alffplay example to support surround sound streams.
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Improved support for building as a sub-project.
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Added an HRTF playback example.
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Added a tone generator output test.
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Added a toolchain to help with cross-compiling to Android.
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openal-soft-1.16.0:
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Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
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effects.
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Implemented high-pass and band-pass EFX filters.
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Implemented the high-pass filter for the EAXReverb effect.
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Implemented SSE2 and SSE4.1 linear resamplers.
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Implemented Neon-enhanced non-HRTF mixers.
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Implemented a QSA backend, for QNX.
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Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
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AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
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extensions.
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Fixed resetting mmdevapi backend devices.
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Fixed clamping when converting 32-bit float samples to integer.
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Fixed modulation range in the Modulator effect.
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Several fixes for the OpenSL playback backend.
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Fixed device specifier names that have Unicode characters on Windows.
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Added support for filenames and paths with Unicode (UTF-8) characters on
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Windows.
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Added support for alsoft.conf config files found in XDG Base Directory
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Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
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defaults) on non-Windows systems.
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Added a GUI configuration utility (requires Qt 4.8).
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Added support for environment variable expansion in config options (not
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keys or section names).
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Added an example that uses SDL2 and ffmpeg.
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Modified examples to use SDL_sound.
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Modified CMake config option names for better sorting.
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HRTF data sets specified in the hrtf_tables config option may now be
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relative or absolute filenames.
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Made the default HRTF data set an external file, and added a data set for
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48khz playback in addition to 44.1khz.
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Added support for C11 atomic methods.
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Improved support for some non-GNU build systems.
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openal-soft-1.15.1:
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Fixed a regression with retrieving the source's AL_GAIN property.
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openal-soft-1.15:
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Fixed device enumeration with the OSS backend.
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Reorganized internal mixing logic, so unneeded steps can potentially be
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skipped for better performance.
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Removed the lookup table for calculating the mixing pans. The panning is
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now calculated directly for better precision.
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Improved the panning of stereo source channels when using stereo output.
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Improved source filter quality on send paths.
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Added a config option to allow PulseAudio to move streams between devices.
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The PulseAudio backend will now attempt to spawn a server by default.
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Added a workaround for a DirectSound bug relating to float32 output.
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Added SSE-based mixers, for HRTF and non-HRTF mixing.
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Added support for the new AL_SOFT_source_latency extension.
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Improved ALSA capture by avoiding an extra buffer when using sizes
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supported by the underlying device.
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Improved the makehrtf utility to support new options and input formats.
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Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
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the header includes can optionally be omitted.
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Added a couple example code programs to show how to apply reverb, and
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retrieve latency.
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The configuration sample is now installed into the share/openal/ directory
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instead of /etc/openal.
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The configuration sample now gets installed by default.
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