3 Implemented the AL_SOFT_UHJ_ex extension.
5 Implemented the AL_SOFT_buffer_length_query extension.
7 Implemented the AL_SOFT_source_start_delay extension.
9 Implemented the AL_EXT_STATIC_BUFFER extension.
11 Fixed compiling with certain older versions of GCC.
13 Fixed compiling as a submodule.
15 Fixed compiling with newer versions of Oboe.
17 Improved EAX effect version switching.
19 Improved the quality of the reverb modulator.
21 Improved performance of the cubic resampler.
23 Added a compatibility option to restore AL_SOFT_buffer_sub_data. The option
24 disables AL_EXT_SOURCE_RADIUS due to incompatibility.
26 Reduced CPU usage when EAX is initialized and FXSlot0 or FXSlot1 are not
29 Reduced memory usage for ADPCM buffer formats. They're no longer converted
30 to 16-bit samples on load.
34 Fixed CoreAudio capture support.
36 Fixed handling per-version EAX properties.
38 Fixed interpolating changes to the Super Stereo width source property.
40 Fixed detection of the update and buffer size from PipeWire.
42 Fixed resuming playback devices with OpenSL.
44 Fixed support for certain OpenAL implementations with the router.
46 Improved reverb environment transitions.
48 Improved performance of convolution reverb.
50 Improved quality and performance of the pitch shifter effect slightly.
52 Improved sub-sample precision for resampled sources.
54 Improved blending spatialized multi-channel sources that use the source
57 Improved mixing 2D ambisonic sources for higher-order 3D ambisonic mixing.
59 Improved quadraphonic and 7.1 surround sound output slightly.
61 Added config options for UHJ encoding/decoding quality. Including Super
64 Added a config option for specifying the speaker distance.
66 Added a compatibility config option for specifying the NFC distance
69 Added a config option for mixing on PipeWire's non-real-time thread.
71 Added support for virtual source nodes with PipeWire capture.
73 Added the ability for the WASAPI backend to use different playback rates.
75 Added support for SOFA files that define per-response delays in makemhr.
77 Changed the default fallback playback sample rate to 48khz. This doesn't
78 affect most backends, which can detect a default rate from the system.
80 Changed the default resampler to cubic.
82 Changed the default HRTF size from 32 to 64 points.
86 Fixed PipeWire version check.
88 Fixed building with PipeWire versions before 0.3.33.
92 Fixed CoreAudio capture.
94 Fixed air absorption strength.
96 Fixed handling 5.1 devices on Windows that use Rear channels instead of
99 Fixed some compilation issues on MinGW.
101 Fixed ALSA not being used on some systems without PipeWire and PulseAudio.
103 Fixed OpenSL capturing noise.
105 Fixed Oboe capture failing with some buffer sizes.
107 Added checks for the runtime PipeWire version. The same or newer version
108 than is used for building will be needed at runtime for the backend to
111 Separated 3D7.1 into its own speaker configuration.
115 Implemented the ALC_SOFT_reopen_device extension. This allows for moving
116 devices to different outputs without losing object state.
118 Implemented the ALC_SOFT_output_mode extension.
120 Implemented the AL_SOFT_callback_buffer extension.
122 Implemented the AL_SOFT_UHJ extension. This supports native UHJ buffer
123 formats and Super Stereo processing.
125 Implemented the legacy EAX extensions. Enabled by default only on Windows.
127 Improved sound positioning stability when a source is near the listener.
129 Improved the default 5.1 output decoder.
131 Improved the high frequency response for the HRTF second-order ambisonic
134 Improved SoundIO capture behavior.
136 Fixed UHJ output on NEON-capable CPUs.
138 Fixed redundant effect updates when setting an effect property to the
141 Fixed WASAPI capture using really low sample rates, and sources with very
142 high pitch shifts when using a bsinc resampler.
144 Added a PipeWire backend.
146 Added enumeration for the JACK and CoreAudio backends.
148 Added optional support for RTKit to get real-time priority. Only used as a
149 backup when pthread_setschedparam fails.
151 Added an option for JACK playback to render directly in the real-time
152 processing callback. For lower playback latency, on by default.
154 Added an option for custom JACK devices.
156 Added utilities to encode and decode UHJ audio files. Files are decoded to
157 the .amb format, and are encoded from libsndfile-compatible formats.
159 Added an in-progress extension to hold sources in a playing state when a
160 device disconnects. Allows devices to be reset or reopened and have sources
161 resume from where they left off.
163 Lowered the priority of the JACK backend. To avoid it getting picked when
164 PipeWire is providing JACK compatibility, since the JACK backend is less
165 robust with auto-configuration.
169 Improved alext.h's detection of standard types.
171 Improved slightly the local source position when the listener and source
174 Improved click/pop prevention for sounds that stop prematurely.
176 Fixed compilation for Windows ARM targets with MSVC.
178 Fixed ARM NEON detection on Windows.
180 Fixed CoreAudio capture when the requested sample rate doesn't match the
181 system configuration.
183 Fixed OpenSL capture desyncing from the internal capture buffer.
185 Fixed sources missing a batch update when applied after quickly restarting
188 Fixed missing source stop events when stopping a paused source.
190 Added capture support to the experimental Oboe backend.
194 Updated library codebase to C++14.
196 Implemented the AL_SOFT_effect_target extension.
198 Implemented the AL_SOFT_events extension.
200 Implemented the ALC_SOFT_loopback_bformat extension.
202 Improved memory use for mixing voices.
204 Improved detection of NEON capabilities.
206 Improved handling of PulseAudio devices that lack manual start control.
208 Improved mixing performance with PulseAudio.
210 Improved high-frequency scaling quality for the HRTF B-Format decoder.
212 Improved makemhr's HRIR delay calculation.
214 Improved WASAPI capture of mono formats with multichannel input.
216 Reimplemented the modulation stage for reverb.
218 Enabled real-time mixing priority by default, for backends that use the
219 setting. It can still be disabled in the config file.
221 Enabled dual-band processing for the built-in quad and 7.1 output decoders.
223 Fixed a potential crash when deleting an effect slot immediately after the
224 last source using it stops.
226 Fixed building with the static runtime on MSVC.
228 Fixed using source stereo angles outside of -pi...+pi.
230 Fixed the buffer processed event count for sources that start with empty
233 Fixed trying to open an unopenable WASAPI device causing all devices to
236 Fixed stale devices when re-enumerating WASAPI devices.
238 Fixed using unicode paths with the log file on Windows.
240 Fixed DirectSound capture reporting bad sample counts or erroring when
243 Added an in-progress extension for a callback-driven buffer type.
245 Added an in-progress extension for higher-order B-Format buffers.
247 Added an in-progress extension for convolution reverb.
249 Added an experimental Oboe backend for Android playback. This requires the
250 Oboe sources at build time, so that it's built as a static library included
253 Added an option for auto-connecting JACK ports.
255 Added greater-than-stereo support to the SoundIO backend.
257 Modified the mixer to be fully asynchronous with the external API, and
258 should now be real-time safe. Although alcRenderSamplesSOFT is not due to
259 locking to check the device handle validity.
261 Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
262 to non-filtered signal phase.
264 Converted examples from SDL_sound to libsndfile. To avoid issues when
265 combining SDL2 and SDL_sound.
267 Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
268 https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
270 Reduced the maximum number of source sends from 16 to 6.
272 Removed the QSA backend. It's been broken for who knows how long.
274 Got rid of the compile-time native-tools targets, using cmake and global
275 initialization instead. This should make cross-compiling less troublesome.
279 Implemented the AL_SOFT_direct_channels_remix extension. This extends
280 AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
281 a matching output channel.
283 Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
284 support for N3D or SN3D scaling, or ACN channel ordering.
286 Fixed a potential voice leak when a source is started and stopped or
287 restarted in quick succession.
289 Fixed a potential device reset failure with JACK.
291 Improved handling of unsupported channel configurations with WASAPI. Such
292 setups will now try to output at least a stereo mix.
294 Improved clarity a bit for the HRTF second-order ambisonic decoder.
296 Improved detection of compatible layouts for SOFA files in makemhr and
299 Added the ability to resample HRTFs on load. MHR files no longer need to
300 match the device sample rate to be usable.
302 Added an option to limit the HRTF's filter length.
306 Converted the library codebase to C++11. A lot of hacks and custom
307 structures have been replaced with standard or cleaner implementations.
309 Partially implemented the Vocal Morpher effect.
311 Fixed the bsinc SSE resamplers on non-GCC compilers.
313 Fixed OpenSL capture.
315 Fixed support for extended capture formats with OpenSL.
317 Fixed handling of WASAPI not reporting a default device.
319 Fixed performance problems relating to semaphores on macOS.
321 Modified the bsinc12 resampler's transition band to better avoid aliasing
324 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
326 Modified the virtual speaker layout for HRTF B-Format decoding.
328 Modified the PulseAudio backend to use a custom processing loop.
330 Renamed the makehrtf utility to makemhr.
332 Improved the efficiency of the bsinc resamplers when up-sampling.
334 Improved the quality of the bsinc resamplers slightly.
336 Improved the efficiency of the HRTF filters.
338 Improved the HRTF B-Format decoder coefficient generation.
340 Improved reverb feedback fading to be more consistent with pan fading.
342 Improved handling of sources that end prematurely, avoiding loud clicks.
344 Improved the performance of some reverb processing loops.
346 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
347 some quality. Notably, down-sampling has less smooth pitch ramping.
349 Added support for SOFA input files with makemhr.
351 Added a build option to use pre-built native tools. For cross-compiling,
352 use with caution and ensure the native tools' binaries are kept up-to-date.
354 Added an adjust-latency config option for the PulseAudio backend.
356 Added basic support for multi-field HRTFs.
358 Added an option for mixing first- or second-order B-Format with HRTF
359 output. This can improve HRTF performance given a number of sources.
361 Added an RC file for proper DLL version information.
363 Disabled some old KDE workarounds by default. Specifically, PulseAudio
364 streams can now be moved (KDE may try to move them after opening).
368 Implemented capture support for the SoundIO backend.
370 Fixed source buffer queues potentially not playing properly when a queue
373 Fixed possible unexpected failures when generating auxiliary effect slots.
375 Fixed a crash with certain reverb or device settings.
377 Fixed OpenSL capture.
379 Improved output limiter response, better ensuring the sample amplitude is
384 Implemented the ALC_SOFT_device_clock extension.
386 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
388 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
390 Fixed compiling on NetBSD.
392 Fixed the reverb effect's density scale and panning parameters.
394 Fixed use of the WASAPI backend with certain games, which caused odd COM
395 initialization errors.
397 Increased the number of virtual channels for decoding Ambisonics to HRTF
400 Changed 32-bit x86 builds to use SSE2 math by default for performance.
401 Build-time options are available to use just SSE1 or x87 instead.
403 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
405 Renamed the MMDevAPI backend to WASAPI.
407 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
408 has been updated to 24-bit.
410 Added a 24- to 48-point band-limited Sinc resampler.
412 Added an SDL2 playback backend. Disabled by default to avoid a dependency
415 Improved the performance and quality of the Chorus and Flanger effects.
417 Improved the efficiency of the band-limited Sinc resampler.
419 Improved the Sinc resampler's transition band to avoid over-attenuating
422 Improved the performance of some filter operations.
424 Improved the efficiency of object ID lookups.
426 Improved the efficienty of internal voice/source synchronization.
428 Improved AL call error logging with contextualized messages.
430 Removed the reverb effect's modulation stage. Due to the lack of reference
431 for its intended behavior and strength.
435 Fixed resetting the FPU rounding mode after certain function calls on
438 Fixed use of SSE intrinsics when building with Clang on Windows.
440 Fixed a crash with the JACK backend when using JACK1.
442 Fixed use of pthread_setnane_np on NetBSD.
444 Fixed building on FreeBSD with an older freebsd-lib.
446 OSS now links with libossaudio if found at build time (for NetBSD).
450 Fixed an issue where resuming a source might not restart playing it.
452 Fixed PulseAudio playback when the configured stream length is much less
453 than the requested length.
455 Fixed MMDevAPI capture with sample rates not matching the backing device.
457 Fixed int32 output for the Wave Writer.
459 Fixed enumeration of OSS devices that are missing device files.
461 Added correct retrieval of the executable's path on FreeBSD.
463 Added a config option to specify the dithering depth.
465 Added a 5.1 decoder preset that excludes front-center output.
469 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
471 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
472 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
474 Implemented 3D processing for some effects. Currently implemented for
475 Reverb, Compressor, Equalizer, and Ring Modulator.
477 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
478 config option to be used.
480 Implemented dual-band processing for high-quality ambisonic decoding.
482 Implemented distance-compensation for surround sound output.
484 Implemented near-field emulation and compensation with ambisonic rendering.
485 Currently only applies when using the high-quality ambisonic decoder or
486 ambisonic output, with appropriate config options.
488 Implemented an output limiter to reduce the amount of distortion from
491 Implemented dithering for 8-bit and 16-bit output.
493 Implemented a config option to select a preferred HRTF.
495 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
497 Implemented experimental capture support for the OpenSL backend.
499 Fixed building on compilers with NEON support but don't default to having
502 Fixed support for JACK on Windows.
504 Fixed starting a source while alcSuspendContext is in effect.
506 Fixed detection of headsets as headphones, with MMDevAPI.
508 Added support for AmbDec config files, for custom ambisonic decoder
509 configurations. Version 3 files only.
511 Added backend-specific options to alsoft-config.
513 Added first-, second-, and third-order ambisonic output formats. Currently
514 only works with backends that don't rely on channel labels, like JACK,
517 Added a build option to embed the default HRTFs into the lib.
519 Added AmbDec presets to enable high-quality ambisonic decoding.
521 Added an AmbDec preset for 3D7.1 speaker setups.
523 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
524 the provided ambdec presets.
526 Added the ability for MMDevAPI to open devices given a Device ID or GUID
529 Added an option to the example apps to open a specific device.
531 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
532 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
535 Increased the default auxiliary effect slot count to 64 (up from 4).
537 Reduced the default period count to 3 (down from 4).
539 Slightly improved automatic naming for enumerated HRTFs.
541 Improved B-Format decoding with HRTF output.
543 Improved internal property handling for better batching behavior.
545 Improved performance of certain filter uses.
547 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
548 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
552 Implemented device enumeration for OSSv4.
554 Fixed building on OSX.
556 Fixed building on non-Windows systems without POSIX-2008.
558 Fixed Dedicated Dialog and Dedicated LFE effect output.
560 Added a build option to override the share install dir.
562 Added a build option to static-link libgcc for MinGW.
566 Fixed building with JACK and without PulseAudio.
568 Fixed building on FreeBSD.
570 Fixed the ALSA backend's allow-resampler option.
572 Fixed handling of inexact ALSA period counts.
574 Altered device naming scheme on Windows backends to better match other
577 Updated the CoreAudio backend to use the AudioComponent API. This clears up
578 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
582 Implemented a JACK playback backend.
584 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
586 Implemented the ALC_SOFT_HRTF extension.
588 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
590 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
591 24-point Sinc resampling, and performs anti-aliasing.
593 Implemented B-Format output support for the wave file writer. This creates
594 FuMa-style first-order Ambisonics wave files (AMB format).
596 Implemented a stereo-mode config option for treating stereo modes as either
597 speakers or headphones.
599 Implemented per-device configuration options.
601 Fixed handling of PulseAudio and MMDevAPI devices that have identical
604 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
606 Fixed logging of Unicode characters on Windows.
608 Fixed 5.1 surround sound channels. By default it will now use the side
609 channels for the surround output. A configuration using rear channels is
612 Fixed the QSA backend potentially altering the capture format.
614 Fixed detecting MMDevAPI's default device.
616 Fixed returning the default capture device name.
618 Fixed mixing property calculations when deferring context updates.
620 Altered the behavior of alcSuspendContext and alcProcessContext to better
621 match certain Windows drivers.
623 Altered the panning algorithm, utilizing Ambisonics for better side and
624 back positioning cues with surround sound output.
626 Improved support for certain older Windows apps.
628 Improved the alffplay example to support surround sound streams.
630 Improved support for building as a sub-project.
632 Added an HRTF playback example.
634 Added a tone generator output test.
636 Added a toolchain to help with cross-compiling to Android.
640 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
643 Implemented high-pass and band-pass EFX filters.
645 Implemented the high-pass filter for the EAXReverb effect.
647 Implemented SSE2 and SSE4.1 linear resamplers.
649 Implemented Neon-enhanced non-HRTF mixers.
651 Implemented a QSA backend, for QNX.
653 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
654 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
657 Fixed resetting mmdevapi backend devices.
659 Fixed clamping when converting 32-bit float samples to integer.
661 Fixed modulation range in the Modulator effect.
663 Several fixes for the OpenSL playback backend.
665 Fixed device specifier names that have Unicode characters on Windows.
667 Added support for filenames and paths with Unicode (UTF-8) characters on
670 Added support for alsoft.conf config files found in XDG Base Directory
671 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
672 defaults) on non-Windows systems.
674 Added a GUI configuration utility (requires Qt 4.8).
676 Added support for environment variable expansion in config options (not
677 keys or section names).
679 Added an example that uses SDL2 and ffmpeg.
681 Modified examples to use SDL_sound.
683 Modified CMake config option names for better sorting.
685 HRTF data sets specified in the hrtf_tables config option may now be
686 relative or absolute filenames.
688 Made the default HRTF data set an external file, and added a data set for
689 48khz playback in addition to 44.1khz.
691 Added support for C11 atomic methods.
693 Improved support for some non-GNU build systems.
697 Fixed a regression with retrieving the source's AL_GAIN property.
701 Fixed device enumeration with the OSS backend.
703 Reorganized internal mixing logic, so unneeded steps can potentially be
704 skipped for better performance.
706 Removed the lookup table for calculating the mixing pans. The panning is
707 now calculated directly for better precision.
709 Improved the panning of stereo source channels when using stereo output.
711 Improved source filter quality on send paths.
713 Added a config option to allow PulseAudio to move streams between devices.
715 The PulseAudio backend will now attempt to spawn a server by default.
717 Added a workaround for a DirectSound bug relating to float32 output.
719 Added SSE-based mixers, for HRTF and non-HRTF mixing.
721 Added support for the new AL_SOFT_source_latency extension.
723 Improved ALSA capture by avoiding an extra buffer when using sizes
724 supported by the underlying device.
726 Improved the makehrtf utility to support new options and input formats.
728 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
729 the header includes can optionally be omitted.
731 Added a couple example code programs to show how to apply reverb, and
734 The configuration sample is now installed into the share/openal/ directory
735 instead of /etc/openal.
737 The configuration sample now gets installed by default.