3 Improved sound positioning when a source is near the listener.
5 Improved the default 5.1 output decoder.
7 Improved the high frequency response for the HRTF second-order ambisonic
10 Improved SoundIO capture behavior.
12 Fixed UHJ output on NEON-capable CPUs.
14 Fixed redundant effect updates when setting an effect property to the
17 Fixed WASAPI capture using really low sample rates, and sources with very
18 high pitch shifts when using a bsinc resampler.
20 Implemented a PipeWire backend.
22 Implemented enumeration for the JACK and CoreAudio backends.
24 Implemented the legacy EAX extension APIs. Enabled by default only on
27 Added optional support for RTKit to get real-time priority. Only used as a
28 backup when pthread_setschedparam fails.
30 Added an option for JACK playback to render directly in the real-time
31 processing callback. For lower playback latency, on by default.
33 Added an option for custom JACK devices.
35 Added utilities to encode and decode UHJ audio files. Files are decoded to
36 the .amb format, and are encoded from libsndfile-compatible formats.
38 Added an in-progress extension to handle UHJ audio buffers natively.
40 Added an in-progress extension to select UHJ output in addition to HRTF.
42 Added an in-progress extension to reopen and move devices to different
43 outputs without losing object state.
45 Added an in-progress extension to hold sources in a playing state when a
46 device disconnects. Allows devices to be reset or reopened and have sources
47 resume from where they left off.
49 Lowered the priority of the JACK backend. To avoid it getting picked when
50 PipeWire is providing JACK compatibility, since the JACK backend is less
51 robust with auto-configuration.
55 Improved alext.h's detection of standard types.
57 Improved slightly the local source position when the listener and source
60 Improved click/pop prevention for sounds that stop prematurely.
62 Fixed compilation for Windows ARM targets with MSVC.
64 Fixed ARM NEON detection on Windows.
66 Fixed CoreAudio capture when the requested sample rate doesn't match the
69 Fixed OpenSL capture desyncing from the internal capture buffer.
71 Fixed sources missing a batch update when applied after quickly restarting
74 Fixed missing source stop events when stopping a paused source.
76 Added capture support to the experimental Oboe backend.
80 Updated library codebase to C++14.
82 Implemented the AL_SOFT_effect_target extension.
84 Implemented the AL_SOFT_events extension.
86 Implemented the ALC_SOFT_loopback_bformat extension.
88 Improved memory use for mixing voices.
90 Improved detection of NEON capabilities.
92 Improved handling of PulseAudio devices that lack manual start control.
94 Improved mixing performance with PulseAudio.
96 Improved high-frequency scaling quality for the HRTF B-Format decoder.
98 Improved makemhr's HRIR delay calculation.
100 Improved WASAPI capture of mono formats with multichannel input.
102 Reimplemented the modulation stage for reverb.
104 Enabled real-time mixing priority by default, for backends that use the
105 setting. It can still be disabled in the config file.
107 Enabled dual-band processing for the built-in quad and 7.1 output decoders.
109 Fixed a potential crash when deleting an effect slot immediately after the
110 last source using it stops.
112 Fixed building with the static runtime on MSVC.
114 Fixed using source stereo angles outside of -pi...+pi.
116 Fixed the buffer processed event count for sources that start with empty
119 Fixed trying to open an unopenable WASAPI device causing all devices to
122 Fixed stale devices when re-enumerating WASAPI devices.
124 Fixed using unicode paths with the log file on Windows.
126 Fixed DirectSound capture reporting bad sample counts or erroring when
129 Added an in-progress extension for a callback-driven buffer type.
131 Added an in-progress extension for higher-order B-Format buffers.
133 Added an in-progress extension for convolution reverb.
135 Added an experimental Oboe backend for Android playback. This requires the
136 Oboe sources at build time, so that it's built as a static library included
139 Added an option for auto-connecting JACK ports.
141 Added greater-than-stereo support to the SoundIO backend.
143 Modified the mixer to be fully asynchronous with the external API, and
144 should now be real-time safe. Although alcRenderSamplesSOFT is not due to
145 locking to check the device handle validity.
147 Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
148 to non-filtered signal phase.
150 Converted examples from SDL_sound to libsndfile. To avoid issues when
151 combining SDL2 and SDL_sound.
153 Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
154 https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
156 Reduced the maximum number of source sends from 16 to 6.
158 Removed the QSA backend. It's been broken for who knows how long.
160 Got rid of the compile-time native-tools targets, using cmake and global
161 initialization instead. This should make cross-compiling less troublesome.
165 Implemented the AL_SOFT_direct_channels_remix extension. This extends
166 AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
167 a matching output channel.
169 Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
170 support for N3D or SN3D scaling, or ACN channel ordering.
172 Fixed a potential voice leak when a source is started and stopped or
173 restarted in quick succession.
175 Fixed a potential device reset failure with JACK.
177 Improved handling of unsupported channel configurations with WASAPI. Such
178 setups will now try to output at least a stereo mix.
180 Improved clarity a bit for the HRTF second-order ambisonic decoder.
182 Improved detection of compatible layouts for SOFA files in makemhr and
185 Added the ability to resample HRTFs on load. MHR files no longer need to
186 match the device sample rate to be usable.
188 Added an option to limit the HRTF's filter length.
192 Converted the library codebase to C++11. A lot of hacks and custom
193 structures have been replaced with standard or cleaner implementations.
195 Partially implemented the Vocal Morpher effect.
197 Fixed the bsinc SSE resamplers on non-GCC compilers.
199 Fixed OpenSL capture.
201 Fixed support for extended capture formats with OpenSL.
203 Fixed handling of WASAPI not reporting a default device.
205 Fixed performance problems relating to semaphores on macOS.
207 Modified the bsinc12 resampler's transition band to better avoid aliasing
210 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
212 Modified the virtual speaker layout for HRTF B-Format decoding.
214 Modified the PulseAudio backend to use a custom processing loop.
216 Renamed the makehrtf utility to makemhr.
218 Improved the efficiency of the bsinc resamplers when up-sampling.
220 Improved the quality of the bsinc resamplers slightly.
222 Improved the efficiency of the HRTF filters.
224 Improved the HRTF B-Format decoder coefficient generation.
226 Improved reverb feedback fading to be more consistent with pan fading.
228 Improved handling of sources that end prematurely, avoiding loud clicks.
230 Improved the performance of some reverb processing loops.
232 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
233 some quality. Notably, down-sampling has less smooth pitch ramping.
235 Added support for SOFA input files with makemhr.
237 Added a build option to use pre-built native tools. For cross-compiling,
238 use with caution and ensure the native tools' binaries are kept up-to-date.
240 Added an adjust-latency config option for the PulseAudio backend.
242 Added basic support for multi-field HRTFs.
244 Added an option for mixing first- or second-order B-Format with HRTF
245 output. This can improve HRTF performance given a number of sources.
247 Added an RC file for proper DLL version information.
249 Disabled some old KDE workarounds by default. Specifically, PulseAudio
250 streams can now be moved (KDE may try to move them after opening).
254 Implemented capture support for the SoundIO backend.
256 Fixed source buffer queues potentially not playing properly when a queue
259 Fixed possible unexpected failures when generating auxiliary effect slots.
261 Fixed a crash with certain reverb or device settings.
263 Fixed OpenSL capture.
265 Improved output limiter response, better ensuring the sample amplitude is
270 Implemented the ALC_SOFT_device_clock extension.
272 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
274 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
276 Fixed compiling on NetBSD.
278 Fixed the reverb effect's density scale and panning parameters.
280 Fixed use of the WASAPI backend with certain games, which caused odd COM
281 initialization errors.
283 Increased the number of virtual channels for decoding Ambisonics to HRTF
286 Changed 32-bit x86 builds to use SSE2 math by default for performance.
287 Build-time options are available to use just SSE1 or x87 instead.
289 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
291 Renamed the MMDevAPI backend to WASAPI.
293 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
294 has been updated to 24-bit.
296 Added a 24- to 48-point band-limited Sinc resampler.
298 Added an SDL2 playback backend. Disabled by default to avoid a dependency
301 Improved the performance and quality of the Chorus and Flanger effects.
303 Improved the efficiency of the band-limited Sinc resampler.
305 Improved the Sinc resampler's transition band to avoid over-attenuating
308 Improved the performance of some filter operations.
310 Improved the efficiency of object ID lookups.
312 Improved the efficienty of internal voice/source synchronization.
314 Improved AL call error logging with contextualized messages.
316 Removed the reverb effect's modulation stage. Due to the lack of reference
317 for its intended behavior and strength.
321 Fixed resetting the FPU rounding mode after certain function calls on
324 Fixed use of SSE intrinsics when building with Clang on Windows.
326 Fixed a crash with the JACK backend when using JACK1.
328 Fixed use of pthread_setnane_np on NetBSD.
330 Fixed building on FreeBSD with an older freebsd-lib.
332 OSS now links with libossaudio if found at build time (for NetBSD).
336 Fixed an issue where resuming a source might not restart playing it.
338 Fixed PulseAudio playback when the configured stream length is much less
339 than the requested length.
341 Fixed MMDevAPI capture with sample rates not matching the backing device.
343 Fixed int32 output for the Wave Writer.
345 Fixed enumeration of OSS devices that are missing device files.
347 Added correct retrieval of the executable's path on FreeBSD.
349 Added a config option to specify the dithering depth.
351 Added a 5.1 decoder preset that excludes front-center output.
355 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
357 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
358 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
360 Implemented 3D processing for some effects. Currently implemented for
361 Reverb, Compressor, Equalizer, and Ring Modulator.
363 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
364 config option to be used.
366 Implemented dual-band processing for high-quality ambisonic decoding.
368 Implemented distance-compensation for surround sound output.
370 Implemented near-field emulation and compensation with ambisonic rendering.
371 Currently only applies when using the high-quality ambisonic decoder or
372 ambisonic output, with appropriate config options.
374 Implemented an output limiter to reduce the amount of distortion from
377 Implemented dithering for 8-bit and 16-bit output.
379 Implemented a config option to select a preferred HRTF.
381 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
383 Implemented experimental capture support for the OpenSL backend.
385 Fixed building on compilers with NEON support but don't default to having
388 Fixed support for JACK on Windows.
390 Fixed starting a source while alcSuspendContext is in effect.
392 Fixed detection of headsets as headphones, with MMDevAPI.
394 Added support for AmbDec config files, for custom ambisonic decoder
395 configurations. Version 3 files only.
397 Added backend-specific options to alsoft-config.
399 Added first-, second-, and third-order ambisonic output formats. Currently
400 only works with backends that don't rely on channel labels, like JACK,
403 Added a build option to embed the default HRTFs into the lib.
405 Added AmbDec presets to enable high-quality ambisonic decoding.
407 Added an AmbDec preset for 3D7.1 speaker setups.
409 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
410 the provided ambdec presets.
412 Added the ability for MMDevAPI to open devices given a Device ID or GUID
415 Added an option to the example apps to open a specific device.
417 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
418 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
421 Increased the default auxiliary effect slot count to 64 (up from 4).
423 Reduced the default period count to 3 (down from 4).
425 Slightly improved automatic naming for enumerated HRTFs.
427 Improved B-Format decoding with HRTF output.
429 Improved internal property handling for better batching behavior.
431 Improved performance of certain filter uses.
433 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
434 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
438 Implemented device enumeration for OSSv4.
440 Fixed building on OSX.
442 Fixed building on non-Windows systems without POSIX-2008.
444 Fixed Dedicated Dialog and Dedicated LFE effect output.
446 Added a build option to override the share install dir.
448 Added a build option to static-link libgcc for MinGW.
452 Fixed building with JACK and without PulseAudio.
454 Fixed building on FreeBSD.
456 Fixed the ALSA backend's allow-resampler option.
458 Fixed handling of inexact ALSA period counts.
460 Altered device naming scheme on Windows backends to better match other
463 Updated the CoreAudio backend to use the AudioComponent API. This clears up
464 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
468 Implemented a JACK playback backend.
470 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
472 Implemented the ALC_SOFT_HRTF extension.
474 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
476 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
477 24-point Sinc resampling, and performs anti-aliasing.
479 Implemented B-Format output support for the wave file writer. This creates
480 FuMa-style first-order Ambisonics wave files (AMB format).
482 Implemented a stereo-mode config option for treating stereo modes as either
483 speakers or headphones.
485 Implemented per-device configuration options.
487 Fixed handling of PulseAudio and MMDevAPI devices that have identical
490 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
492 Fixed logging of Unicode characters on Windows.
494 Fixed 5.1 surround sound channels. By default it will now use the side
495 channels for the surround output. A configuration using rear channels is
498 Fixed the QSA backend potentially altering the capture format.
500 Fixed detecting MMDevAPI's default device.
502 Fixed returning the default capture device name.
504 Fixed mixing property calculations when deferring context updates.
506 Altered the behavior of alcSuspendContext and alcProcessContext to better
507 match certain Windows drivers.
509 Altered the panning algorithm, utilizing Ambisonics for better side and
510 back positioning cues with surround sound output.
512 Improved support for certain older Windows apps.
514 Improved the alffplay example to support surround sound streams.
516 Improved support for building as a sub-project.
518 Added an HRTF playback example.
520 Added a tone generator output test.
522 Added a toolchain to help with cross-compiling to Android.
526 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
529 Implemented high-pass and band-pass EFX filters.
531 Implemented the high-pass filter for the EAXReverb effect.
533 Implemented SSE2 and SSE4.1 linear resamplers.
535 Implemented Neon-enhanced non-HRTF mixers.
537 Implemented a QSA backend, for QNX.
539 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
540 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
543 Fixed resetting mmdevapi backend devices.
545 Fixed clamping when converting 32-bit float samples to integer.
547 Fixed modulation range in the Modulator effect.
549 Several fixes for the OpenSL playback backend.
551 Fixed device specifier names that have Unicode characters on Windows.
553 Added support for filenames and paths with Unicode (UTF-8) characters on
556 Added support for alsoft.conf config files found in XDG Base Directory
557 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
558 defaults) on non-Windows systems.
560 Added a GUI configuration utility (requires Qt 4.8).
562 Added support for environment variable expansion in config options (not
563 keys or section names).
565 Added an example that uses SDL2 and ffmpeg.
567 Modified examples to use SDL_sound.
569 Modified CMake config option names for better sorting.
571 HRTF data sets specified in the hrtf_tables config option may now be
572 relative or absolute filenames.
574 Made the default HRTF data set an external file, and added a data set for
575 48khz playback in addition to 44.1khz.
577 Added support for C11 atomic methods.
579 Improved support for some non-GNU build systems.
583 Fixed a regression with retrieving the source's AL_GAIN property.
587 Fixed device enumeration with the OSS backend.
589 Reorganized internal mixing logic, so unneeded steps can potentially be
590 skipped for better performance.
592 Removed the lookup table for calculating the mixing pans. The panning is
593 now calculated directly for better precision.
595 Improved the panning of stereo source channels when using stereo output.
597 Improved source filter quality on send paths.
599 Added a config option to allow PulseAudio to move streams between devices.
601 The PulseAudio backend will now attempt to spawn a server by default.
603 Added a workaround for a DirectSound bug relating to float32 output.
605 Added SSE-based mixers, for HRTF and non-HRTF mixing.
607 Added support for the new AL_SOFT_source_latency extension.
609 Improved ALSA capture by avoiding an extra buffer when using sizes
610 supported by the underlying device.
612 Improved the makehrtf utility to support new options and input formats.
614 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
615 the header includes can optionally be omitted.
617 Added a couple example code programs to show how to apply reverb, and
620 The configuration sample is now installed into the share/openal/ directory
621 instead of /etc/openal.
623 The configuration sample now gets installed by default.