3 Fixed PipeWire version check.
5 Fixed building with PipeWire versions before 0.3.33.
9 Fixed CoreAudio capture.
11 Fixed air absorption strength.
13 Fixed handling 5.1 devices on Windows that use Rear channels instead of
16 Fixed some compilation issues on MinGW.
18 Fixed ALSA not being used on some systems without PipeWire and PulseAudio.
20 Fixed OpenSL capturing noise.
22 Fixed Oboe capture failing with some buffer sizes.
24 Added checks for the runtime PipeWire version. The same or newer version
25 than is used for building will be needed at runtime for the backend to
28 Separated 3D7.1 into its own speaker configuration.
32 Implemented the ALC_SOFT_reopen_device extension. This allows for moving
33 devices to different outputs without losing object state.
35 Implemented the ALC_SOFT_output_mode extension.
37 Implemented the AL_SOFT_callback_buffer extension.
39 Implemented the AL_SOFT_UHJ extension. This supports native UHJ buffer
40 formats and Super Stereo processing.
42 Implemented the legacy EAX extensions. Enabled by default only on Windows.
44 Improved sound positioning stability when a source is near the listener.
46 Improved the default 5.1 output decoder.
48 Improved the high frequency response for the HRTF second-order ambisonic
51 Improved SoundIO capture behavior.
53 Fixed UHJ output on NEON-capable CPUs.
55 Fixed redundant effect updates when setting an effect property to the
58 Fixed WASAPI capture using really low sample rates, and sources with very
59 high pitch shifts when using a bsinc resampler.
61 Added a PipeWire backend.
63 Added enumeration for the JACK and CoreAudio backends.
65 Added optional support for RTKit to get real-time priority. Only used as a
66 backup when pthread_setschedparam fails.
68 Added an option for JACK playback to render directly in the real-time
69 processing callback. For lower playback latency, on by default.
71 Added an option for custom JACK devices.
73 Added utilities to encode and decode UHJ audio files. Files are decoded to
74 the .amb format, and are encoded from libsndfile-compatible formats.
76 Added an in-progress extension to hold sources in a playing state when a
77 device disconnects. Allows devices to be reset or reopened and have sources
78 resume from where they left off.
80 Lowered the priority of the JACK backend. To avoid it getting picked when
81 PipeWire is providing JACK compatibility, since the JACK backend is less
82 robust with auto-configuration.
86 Improved alext.h's detection of standard types.
88 Improved slightly the local source position when the listener and source
91 Improved click/pop prevention for sounds that stop prematurely.
93 Fixed compilation for Windows ARM targets with MSVC.
95 Fixed ARM NEON detection on Windows.
97 Fixed CoreAudio capture when the requested sample rate doesn't match the
100 Fixed OpenSL capture desyncing from the internal capture buffer.
102 Fixed sources missing a batch update when applied after quickly restarting
105 Fixed missing source stop events when stopping a paused source.
107 Added capture support to the experimental Oboe backend.
111 Updated library codebase to C++14.
113 Implemented the AL_SOFT_effect_target extension.
115 Implemented the AL_SOFT_events extension.
117 Implemented the ALC_SOFT_loopback_bformat extension.
119 Improved memory use for mixing voices.
121 Improved detection of NEON capabilities.
123 Improved handling of PulseAudio devices that lack manual start control.
125 Improved mixing performance with PulseAudio.
127 Improved high-frequency scaling quality for the HRTF B-Format decoder.
129 Improved makemhr's HRIR delay calculation.
131 Improved WASAPI capture of mono formats with multichannel input.
133 Reimplemented the modulation stage for reverb.
135 Enabled real-time mixing priority by default, for backends that use the
136 setting. It can still be disabled in the config file.
138 Enabled dual-band processing for the built-in quad and 7.1 output decoders.
140 Fixed a potential crash when deleting an effect slot immediately after the
141 last source using it stops.
143 Fixed building with the static runtime on MSVC.
145 Fixed using source stereo angles outside of -pi...+pi.
147 Fixed the buffer processed event count for sources that start with empty
150 Fixed trying to open an unopenable WASAPI device causing all devices to
153 Fixed stale devices when re-enumerating WASAPI devices.
155 Fixed using unicode paths with the log file on Windows.
157 Fixed DirectSound capture reporting bad sample counts or erroring when
160 Added an in-progress extension for a callback-driven buffer type.
162 Added an in-progress extension for higher-order B-Format buffers.
164 Added an in-progress extension for convolution reverb.
166 Added an experimental Oboe backend for Android playback. This requires the
167 Oboe sources at build time, so that it's built as a static library included
170 Added an option for auto-connecting JACK ports.
172 Added greater-than-stereo support to the SoundIO backend.
174 Modified the mixer to be fully asynchronous with the external API, and
175 should now be real-time safe. Although alcRenderSamplesSOFT is not due to
176 locking to check the device handle validity.
178 Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
179 to non-filtered signal phase.
181 Converted examples from SDL_sound to libsndfile. To avoid issues when
182 combining SDL2 and SDL_sound.
184 Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
185 https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
187 Reduced the maximum number of source sends from 16 to 6.
189 Removed the QSA backend. It's been broken for who knows how long.
191 Got rid of the compile-time native-tools targets, using cmake and global
192 initialization instead. This should make cross-compiling less troublesome.
196 Implemented the AL_SOFT_direct_channels_remix extension. This extends
197 AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
198 a matching output channel.
200 Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
201 support for N3D or SN3D scaling, or ACN channel ordering.
203 Fixed a potential voice leak when a source is started and stopped or
204 restarted in quick succession.
206 Fixed a potential device reset failure with JACK.
208 Improved handling of unsupported channel configurations with WASAPI. Such
209 setups will now try to output at least a stereo mix.
211 Improved clarity a bit for the HRTF second-order ambisonic decoder.
213 Improved detection of compatible layouts for SOFA files in makemhr and
216 Added the ability to resample HRTFs on load. MHR files no longer need to
217 match the device sample rate to be usable.
219 Added an option to limit the HRTF's filter length.
223 Converted the library codebase to C++11. A lot of hacks and custom
224 structures have been replaced with standard or cleaner implementations.
226 Partially implemented the Vocal Morpher effect.
228 Fixed the bsinc SSE resamplers on non-GCC compilers.
230 Fixed OpenSL capture.
232 Fixed support for extended capture formats with OpenSL.
234 Fixed handling of WASAPI not reporting a default device.
236 Fixed performance problems relating to semaphores on macOS.
238 Modified the bsinc12 resampler's transition band to better avoid aliasing
241 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
243 Modified the virtual speaker layout for HRTF B-Format decoding.
245 Modified the PulseAudio backend to use a custom processing loop.
247 Renamed the makehrtf utility to makemhr.
249 Improved the efficiency of the bsinc resamplers when up-sampling.
251 Improved the quality of the bsinc resamplers slightly.
253 Improved the efficiency of the HRTF filters.
255 Improved the HRTF B-Format decoder coefficient generation.
257 Improved reverb feedback fading to be more consistent with pan fading.
259 Improved handling of sources that end prematurely, avoiding loud clicks.
261 Improved the performance of some reverb processing loops.
263 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
264 some quality. Notably, down-sampling has less smooth pitch ramping.
266 Added support for SOFA input files with makemhr.
268 Added a build option to use pre-built native tools. For cross-compiling,
269 use with caution and ensure the native tools' binaries are kept up-to-date.
271 Added an adjust-latency config option for the PulseAudio backend.
273 Added basic support for multi-field HRTFs.
275 Added an option for mixing first- or second-order B-Format with HRTF
276 output. This can improve HRTF performance given a number of sources.
278 Added an RC file for proper DLL version information.
280 Disabled some old KDE workarounds by default. Specifically, PulseAudio
281 streams can now be moved (KDE may try to move them after opening).
285 Implemented capture support for the SoundIO backend.
287 Fixed source buffer queues potentially not playing properly when a queue
290 Fixed possible unexpected failures when generating auxiliary effect slots.
292 Fixed a crash with certain reverb or device settings.
294 Fixed OpenSL capture.
296 Improved output limiter response, better ensuring the sample amplitude is
301 Implemented the ALC_SOFT_device_clock extension.
303 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
305 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
307 Fixed compiling on NetBSD.
309 Fixed the reverb effect's density scale and panning parameters.
311 Fixed use of the WASAPI backend with certain games, which caused odd COM
312 initialization errors.
314 Increased the number of virtual channels for decoding Ambisonics to HRTF
317 Changed 32-bit x86 builds to use SSE2 math by default for performance.
318 Build-time options are available to use just SSE1 or x87 instead.
320 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
322 Renamed the MMDevAPI backend to WASAPI.
324 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
325 has been updated to 24-bit.
327 Added a 24- to 48-point band-limited Sinc resampler.
329 Added an SDL2 playback backend. Disabled by default to avoid a dependency
332 Improved the performance and quality of the Chorus and Flanger effects.
334 Improved the efficiency of the band-limited Sinc resampler.
336 Improved the Sinc resampler's transition band to avoid over-attenuating
339 Improved the performance of some filter operations.
341 Improved the efficiency of object ID lookups.
343 Improved the efficienty of internal voice/source synchronization.
345 Improved AL call error logging with contextualized messages.
347 Removed the reverb effect's modulation stage. Due to the lack of reference
348 for its intended behavior and strength.
352 Fixed resetting the FPU rounding mode after certain function calls on
355 Fixed use of SSE intrinsics when building with Clang on Windows.
357 Fixed a crash with the JACK backend when using JACK1.
359 Fixed use of pthread_setnane_np on NetBSD.
361 Fixed building on FreeBSD with an older freebsd-lib.
363 OSS now links with libossaudio if found at build time (for NetBSD).
367 Fixed an issue where resuming a source might not restart playing it.
369 Fixed PulseAudio playback when the configured stream length is much less
370 than the requested length.
372 Fixed MMDevAPI capture with sample rates not matching the backing device.
374 Fixed int32 output for the Wave Writer.
376 Fixed enumeration of OSS devices that are missing device files.
378 Added correct retrieval of the executable's path on FreeBSD.
380 Added a config option to specify the dithering depth.
382 Added a 5.1 decoder preset that excludes front-center output.
386 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
388 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
389 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
391 Implemented 3D processing for some effects. Currently implemented for
392 Reverb, Compressor, Equalizer, and Ring Modulator.
394 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
395 config option to be used.
397 Implemented dual-band processing for high-quality ambisonic decoding.
399 Implemented distance-compensation for surround sound output.
401 Implemented near-field emulation and compensation with ambisonic rendering.
402 Currently only applies when using the high-quality ambisonic decoder or
403 ambisonic output, with appropriate config options.
405 Implemented an output limiter to reduce the amount of distortion from
408 Implemented dithering for 8-bit and 16-bit output.
410 Implemented a config option to select a preferred HRTF.
412 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
414 Implemented experimental capture support for the OpenSL backend.
416 Fixed building on compilers with NEON support but don't default to having
419 Fixed support for JACK on Windows.
421 Fixed starting a source while alcSuspendContext is in effect.
423 Fixed detection of headsets as headphones, with MMDevAPI.
425 Added support for AmbDec config files, for custom ambisonic decoder
426 configurations. Version 3 files only.
428 Added backend-specific options to alsoft-config.
430 Added first-, second-, and third-order ambisonic output formats. Currently
431 only works with backends that don't rely on channel labels, like JACK,
434 Added a build option to embed the default HRTFs into the lib.
436 Added AmbDec presets to enable high-quality ambisonic decoding.
438 Added an AmbDec preset for 3D7.1 speaker setups.
440 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
441 the provided ambdec presets.
443 Added the ability for MMDevAPI to open devices given a Device ID or GUID
446 Added an option to the example apps to open a specific device.
448 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
449 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
452 Increased the default auxiliary effect slot count to 64 (up from 4).
454 Reduced the default period count to 3 (down from 4).
456 Slightly improved automatic naming for enumerated HRTFs.
458 Improved B-Format decoding with HRTF output.
460 Improved internal property handling for better batching behavior.
462 Improved performance of certain filter uses.
464 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
465 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
469 Implemented device enumeration for OSSv4.
471 Fixed building on OSX.
473 Fixed building on non-Windows systems without POSIX-2008.
475 Fixed Dedicated Dialog and Dedicated LFE effect output.
477 Added a build option to override the share install dir.
479 Added a build option to static-link libgcc for MinGW.
483 Fixed building with JACK and without PulseAudio.
485 Fixed building on FreeBSD.
487 Fixed the ALSA backend's allow-resampler option.
489 Fixed handling of inexact ALSA period counts.
491 Altered device naming scheme on Windows backends to better match other
494 Updated the CoreAudio backend to use the AudioComponent API. This clears up
495 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
499 Implemented a JACK playback backend.
501 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
503 Implemented the ALC_SOFT_HRTF extension.
505 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
507 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
508 24-point Sinc resampling, and performs anti-aliasing.
510 Implemented B-Format output support for the wave file writer. This creates
511 FuMa-style first-order Ambisonics wave files (AMB format).
513 Implemented a stereo-mode config option for treating stereo modes as either
514 speakers or headphones.
516 Implemented per-device configuration options.
518 Fixed handling of PulseAudio and MMDevAPI devices that have identical
521 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
523 Fixed logging of Unicode characters on Windows.
525 Fixed 5.1 surround sound channels. By default it will now use the side
526 channels for the surround output. A configuration using rear channels is
529 Fixed the QSA backend potentially altering the capture format.
531 Fixed detecting MMDevAPI's default device.
533 Fixed returning the default capture device name.
535 Fixed mixing property calculations when deferring context updates.
537 Altered the behavior of alcSuspendContext and alcProcessContext to better
538 match certain Windows drivers.
540 Altered the panning algorithm, utilizing Ambisonics for better side and
541 back positioning cues with surround sound output.
543 Improved support for certain older Windows apps.
545 Improved the alffplay example to support surround sound streams.
547 Improved support for building as a sub-project.
549 Added an HRTF playback example.
551 Added a tone generator output test.
553 Added a toolchain to help with cross-compiling to Android.
557 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
560 Implemented high-pass and band-pass EFX filters.
562 Implemented the high-pass filter for the EAXReverb effect.
564 Implemented SSE2 and SSE4.1 linear resamplers.
566 Implemented Neon-enhanced non-HRTF mixers.
568 Implemented a QSA backend, for QNX.
570 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
571 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
574 Fixed resetting mmdevapi backend devices.
576 Fixed clamping when converting 32-bit float samples to integer.
578 Fixed modulation range in the Modulator effect.
580 Several fixes for the OpenSL playback backend.
582 Fixed device specifier names that have Unicode characters on Windows.
584 Added support for filenames and paths with Unicode (UTF-8) characters on
587 Added support for alsoft.conf config files found in XDG Base Directory
588 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
589 defaults) on non-Windows systems.
591 Added a GUI configuration utility (requires Qt 4.8).
593 Added support for environment variable expansion in config options (not
594 keys or section names).
596 Added an example that uses SDL2 and ffmpeg.
598 Modified examples to use SDL_sound.
600 Modified CMake config option names for better sorting.
602 HRTF data sets specified in the hrtf_tables config option may now be
603 relative or absolute filenames.
605 Made the default HRTF data set an external file, and added a data set for
606 48khz playback in addition to 44.1khz.
608 Added support for C11 atomic methods.
610 Improved support for some non-GNU build systems.
614 Fixed a regression with retrieving the source's AL_GAIN property.
618 Fixed device enumeration with the OSS backend.
620 Reorganized internal mixing logic, so unneeded steps can potentially be
621 skipped for better performance.
623 Removed the lookup table for calculating the mixing pans. The panning is
624 now calculated directly for better precision.
626 Improved the panning of stereo source channels when using stereo output.
628 Improved source filter quality on send paths.
630 Added a config option to allow PulseAudio to move streams between devices.
632 The PulseAudio backend will now attempt to spawn a server by default.
634 Added a workaround for a DirectSound bug relating to float32 output.
636 Added SSE-based mixers, for HRTF and non-HRTF mixing.
638 Added support for the new AL_SOFT_source_latency extension.
640 Improved ALSA capture by avoiding an extra buffer when using sizes
641 supported by the underlying device.
643 Improved the makehrtf utility to support new options and input formats.
645 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
646 the header includes can optionally be omitted.
648 Added a couple example code programs to show how to apply reverb, and
651 The configuration sample is now installed into the share/openal/ directory
652 instead of /etc/openal.
654 The configuration sample now gets installed by default.