3 Fixed CoreAudio capture.
5 Fixed air absorption strength.
7 Fixed silent surround speakers with certain 5.1 devices on Windows.
9 Fixed some compilation issues on MinGW.
11 Added checks for the runtime PipeWire version. The same or newer version
12 than is used for building will be needed at runtime for the backend to
15 Separated 3D7.1 into its own speaker configuration.
19 Implemented the ALC_SOFT_reopen_device extension. This allows for moving
20 devices to different outputs without losing object state.
22 Implemented the ALC_SOFT_output_mode extension.
24 Implemented the AL_SOFT_callback_buffer extension.
26 Implemented the AL_SOFT_UHJ extension. This supports native UHJ buffer
27 formats and Super Stereo processing.
29 Implemented the legacy EAX extensions. Enabled by default only on Windows.
31 Improved sound positioning stability when a source is near the listener.
33 Improved the default 5.1 output decoder.
35 Improved the high frequency response for the HRTF second-order ambisonic
38 Improved SoundIO capture behavior.
40 Fixed UHJ output on NEON-capable CPUs.
42 Fixed redundant effect updates when setting an effect property to the
45 Fixed WASAPI capture using really low sample rates, and sources with very
46 high pitch shifts when using a bsinc resampler.
48 Added a PipeWire backend.
50 Added enumeration for the JACK and CoreAudio backends.
52 Added optional support for RTKit to get real-time priority. Only used as a
53 backup when pthread_setschedparam fails.
55 Added an option for JACK playback to render directly in the real-time
56 processing callback. For lower playback latency, on by default.
58 Added an option for custom JACK devices.
60 Added utilities to encode and decode UHJ audio files. Files are decoded to
61 the .amb format, and are encoded from libsndfile-compatible formats.
63 Added an in-progress extension to hold sources in a playing state when a
64 device disconnects. Allows devices to be reset or reopened and have sources
65 resume from where they left off.
67 Lowered the priority of the JACK backend. To avoid it getting picked when
68 PipeWire is providing JACK compatibility, since the JACK backend is less
69 robust with auto-configuration.
73 Improved alext.h's detection of standard types.
75 Improved slightly the local source position when the listener and source
78 Improved click/pop prevention for sounds that stop prematurely.
80 Fixed compilation for Windows ARM targets with MSVC.
82 Fixed ARM NEON detection on Windows.
84 Fixed CoreAudio capture when the requested sample rate doesn't match the
87 Fixed OpenSL capture desyncing from the internal capture buffer.
89 Fixed sources missing a batch update when applied after quickly restarting
92 Fixed missing source stop events when stopping a paused source.
94 Added capture support to the experimental Oboe backend.
98 Updated library codebase to C++14.
100 Implemented the AL_SOFT_effect_target extension.
102 Implemented the AL_SOFT_events extension.
104 Implemented the ALC_SOFT_loopback_bformat extension.
106 Improved memory use for mixing voices.
108 Improved detection of NEON capabilities.
110 Improved handling of PulseAudio devices that lack manual start control.
112 Improved mixing performance with PulseAudio.
114 Improved high-frequency scaling quality for the HRTF B-Format decoder.
116 Improved makemhr's HRIR delay calculation.
118 Improved WASAPI capture of mono formats with multichannel input.
120 Reimplemented the modulation stage for reverb.
122 Enabled real-time mixing priority by default, for backends that use the
123 setting. It can still be disabled in the config file.
125 Enabled dual-band processing for the built-in quad and 7.1 output decoders.
127 Fixed a potential crash when deleting an effect slot immediately after the
128 last source using it stops.
130 Fixed building with the static runtime on MSVC.
132 Fixed using source stereo angles outside of -pi...+pi.
134 Fixed the buffer processed event count for sources that start with empty
137 Fixed trying to open an unopenable WASAPI device causing all devices to
140 Fixed stale devices when re-enumerating WASAPI devices.
142 Fixed using unicode paths with the log file on Windows.
144 Fixed DirectSound capture reporting bad sample counts or erroring when
147 Added an in-progress extension for a callback-driven buffer type.
149 Added an in-progress extension for higher-order B-Format buffers.
151 Added an in-progress extension for convolution reverb.
153 Added an experimental Oboe backend for Android playback. This requires the
154 Oboe sources at build time, so that it's built as a static library included
157 Added an option for auto-connecting JACK ports.
159 Added greater-than-stereo support to the SoundIO backend.
161 Modified the mixer to be fully asynchronous with the external API, and
162 should now be real-time safe. Although alcRenderSamplesSOFT is not due to
163 locking to check the device handle validity.
165 Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
166 to non-filtered signal phase.
168 Converted examples from SDL_sound to libsndfile. To avoid issues when
169 combining SDL2 and SDL_sound.
171 Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
172 https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
174 Reduced the maximum number of source sends from 16 to 6.
176 Removed the QSA backend. It's been broken for who knows how long.
178 Got rid of the compile-time native-tools targets, using cmake and global
179 initialization instead. This should make cross-compiling less troublesome.
183 Implemented the AL_SOFT_direct_channels_remix extension. This extends
184 AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
185 a matching output channel.
187 Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
188 support for N3D or SN3D scaling, or ACN channel ordering.
190 Fixed a potential voice leak when a source is started and stopped or
191 restarted in quick succession.
193 Fixed a potential device reset failure with JACK.
195 Improved handling of unsupported channel configurations with WASAPI. Such
196 setups will now try to output at least a stereo mix.
198 Improved clarity a bit for the HRTF second-order ambisonic decoder.
200 Improved detection of compatible layouts for SOFA files in makemhr and
203 Added the ability to resample HRTFs on load. MHR files no longer need to
204 match the device sample rate to be usable.
206 Added an option to limit the HRTF's filter length.
210 Converted the library codebase to C++11. A lot of hacks and custom
211 structures have been replaced with standard or cleaner implementations.
213 Partially implemented the Vocal Morpher effect.
215 Fixed the bsinc SSE resamplers on non-GCC compilers.
217 Fixed OpenSL capture.
219 Fixed support for extended capture formats with OpenSL.
221 Fixed handling of WASAPI not reporting a default device.
223 Fixed performance problems relating to semaphores on macOS.
225 Modified the bsinc12 resampler's transition band to better avoid aliasing
228 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
230 Modified the virtual speaker layout for HRTF B-Format decoding.
232 Modified the PulseAudio backend to use a custom processing loop.
234 Renamed the makehrtf utility to makemhr.
236 Improved the efficiency of the bsinc resamplers when up-sampling.
238 Improved the quality of the bsinc resamplers slightly.
240 Improved the efficiency of the HRTF filters.
242 Improved the HRTF B-Format decoder coefficient generation.
244 Improved reverb feedback fading to be more consistent with pan fading.
246 Improved handling of sources that end prematurely, avoiding loud clicks.
248 Improved the performance of some reverb processing loops.
250 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
251 some quality. Notably, down-sampling has less smooth pitch ramping.
253 Added support for SOFA input files with makemhr.
255 Added a build option to use pre-built native tools. For cross-compiling,
256 use with caution and ensure the native tools' binaries are kept up-to-date.
258 Added an adjust-latency config option for the PulseAudio backend.
260 Added basic support for multi-field HRTFs.
262 Added an option for mixing first- or second-order B-Format with HRTF
263 output. This can improve HRTF performance given a number of sources.
265 Added an RC file for proper DLL version information.
267 Disabled some old KDE workarounds by default. Specifically, PulseAudio
268 streams can now be moved (KDE may try to move them after opening).
272 Implemented capture support for the SoundIO backend.
274 Fixed source buffer queues potentially not playing properly when a queue
277 Fixed possible unexpected failures when generating auxiliary effect slots.
279 Fixed a crash with certain reverb or device settings.
281 Fixed OpenSL capture.
283 Improved output limiter response, better ensuring the sample amplitude is
288 Implemented the ALC_SOFT_device_clock extension.
290 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
292 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
294 Fixed compiling on NetBSD.
296 Fixed the reverb effect's density scale and panning parameters.
298 Fixed use of the WASAPI backend with certain games, which caused odd COM
299 initialization errors.
301 Increased the number of virtual channels for decoding Ambisonics to HRTF
304 Changed 32-bit x86 builds to use SSE2 math by default for performance.
305 Build-time options are available to use just SSE1 or x87 instead.
307 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
309 Renamed the MMDevAPI backend to WASAPI.
311 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
312 has been updated to 24-bit.
314 Added a 24- to 48-point band-limited Sinc resampler.
316 Added an SDL2 playback backend. Disabled by default to avoid a dependency
319 Improved the performance and quality of the Chorus and Flanger effects.
321 Improved the efficiency of the band-limited Sinc resampler.
323 Improved the Sinc resampler's transition band to avoid over-attenuating
326 Improved the performance of some filter operations.
328 Improved the efficiency of object ID lookups.
330 Improved the efficienty of internal voice/source synchronization.
332 Improved AL call error logging with contextualized messages.
334 Removed the reverb effect's modulation stage. Due to the lack of reference
335 for its intended behavior and strength.
339 Fixed resetting the FPU rounding mode after certain function calls on
342 Fixed use of SSE intrinsics when building with Clang on Windows.
344 Fixed a crash with the JACK backend when using JACK1.
346 Fixed use of pthread_setnane_np on NetBSD.
348 Fixed building on FreeBSD with an older freebsd-lib.
350 OSS now links with libossaudio if found at build time (for NetBSD).
354 Fixed an issue where resuming a source might not restart playing it.
356 Fixed PulseAudio playback when the configured stream length is much less
357 than the requested length.
359 Fixed MMDevAPI capture with sample rates not matching the backing device.
361 Fixed int32 output for the Wave Writer.
363 Fixed enumeration of OSS devices that are missing device files.
365 Added correct retrieval of the executable's path on FreeBSD.
367 Added a config option to specify the dithering depth.
369 Added a 5.1 decoder preset that excludes front-center output.
373 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
375 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
376 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
378 Implemented 3D processing for some effects. Currently implemented for
379 Reverb, Compressor, Equalizer, and Ring Modulator.
381 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
382 config option to be used.
384 Implemented dual-band processing for high-quality ambisonic decoding.
386 Implemented distance-compensation for surround sound output.
388 Implemented near-field emulation and compensation with ambisonic rendering.
389 Currently only applies when using the high-quality ambisonic decoder or
390 ambisonic output, with appropriate config options.
392 Implemented an output limiter to reduce the amount of distortion from
395 Implemented dithering for 8-bit and 16-bit output.
397 Implemented a config option to select a preferred HRTF.
399 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
401 Implemented experimental capture support for the OpenSL backend.
403 Fixed building on compilers with NEON support but don't default to having
406 Fixed support for JACK on Windows.
408 Fixed starting a source while alcSuspendContext is in effect.
410 Fixed detection of headsets as headphones, with MMDevAPI.
412 Added support for AmbDec config files, for custom ambisonic decoder
413 configurations. Version 3 files only.
415 Added backend-specific options to alsoft-config.
417 Added first-, second-, and third-order ambisonic output formats. Currently
418 only works with backends that don't rely on channel labels, like JACK,
421 Added a build option to embed the default HRTFs into the lib.
423 Added AmbDec presets to enable high-quality ambisonic decoding.
425 Added an AmbDec preset for 3D7.1 speaker setups.
427 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
428 the provided ambdec presets.
430 Added the ability for MMDevAPI to open devices given a Device ID or GUID
433 Added an option to the example apps to open a specific device.
435 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
436 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
439 Increased the default auxiliary effect slot count to 64 (up from 4).
441 Reduced the default period count to 3 (down from 4).
443 Slightly improved automatic naming for enumerated HRTFs.
445 Improved B-Format decoding with HRTF output.
447 Improved internal property handling for better batching behavior.
449 Improved performance of certain filter uses.
451 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
452 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
456 Implemented device enumeration for OSSv4.
458 Fixed building on OSX.
460 Fixed building on non-Windows systems without POSIX-2008.
462 Fixed Dedicated Dialog and Dedicated LFE effect output.
464 Added a build option to override the share install dir.
466 Added a build option to static-link libgcc for MinGW.
470 Fixed building with JACK and without PulseAudio.
472 Fixed building on FreeBSD.
474 Fixed the ALSA backend's allow-resampler option.
476 Fixed handling of inexact ALSA period counts.
478 Altered device naming scheme on Windows backends to better match other
481 Updated the CoreAudio backend to use the AudioComponent API. This clears up
482 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
486 Implemented a JACK playback backend.
488 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
490 Implemented the ALC_SOFT_HRTF extension.
492 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
494 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
495 24-point Sinc resampling, and performs anti-aliasing.
497 Implemented B-Format output support for the wave file writer. This creates
498 FuMa-style first-order Ambisonics wave files (AMB format).
500 Implemented a stereo-mode config option for treating stereo modes as either
501 speakers or headphones.
503 Implemented per-device configuration options.
505 Fixed handling of PulseAudio and MMDevAPI devices that have identical
508 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
510 Fixed logging of Unicode characters on Windows.
512 Fixed 5.1 surround sound channels. By default it will now use the side
513 channels for the surround output. A configuration using rear channels is
516 Fixed the QSA backend potentially altering the capture format.
518 Fixed detecting MMDevAPI's default device.
520 Fixed returning the default capture device name.
522 Fixed mixing property calculations when deferring context updates.
524 Altered the behavior of alcSuspendContext and alcProcessContext to better
525 match certain Windows drivers.
527 Altered the panning algorithm, utilizing Ambisonics for better side and
528 back positioning cues with surround sound output.
530 Improved support for certain older Windows apps.
532 Improved the alffplay example to support surround sound streams.
534 Improved support for building as a sub-project.
536 Added an HRTF playback example.
538 Added a tone generator output test.
540 Added a toolchain to help with cross-compiling to Android.
544 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
547 Implemented high-pass and band-pass EFX filters.
549 Implemented the high-pass filter for the EAXReverb effect.
551 Implemented SSE2 and SSE4.1 linear resamplers.
553 Implemented Neon-enhanced non-HRTF mixers.
555 Implemented a QSA backend, for QNX.
557 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
558 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
561 Fixed resetting mmdevapi backend devices.
563 Fixed clamping when converting 32-bit float samples to integer.
565 Fixed modulation range in the Modulator effect.
567 Several fixes for the OpenSL playback backend.
569 Fixed device specifier names that have Unicode characters on Windows.
571 Added support for filenames and paths with Unicode (UTF-8) characters on
574 Added support for alsoft.conf config files found in XDG Base Directory
575 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
576 defaults) on non-Windows systems.
578 Added a GUI configuration utility (requires Qt 4.8).
580 Added support for environment variable expansion in config options (not
581 keys or section names).
583 Added an example that uses SDL2 and ffmpeg.
585 Modified examples to use SDL_sound.
587 Modified CMake config option names for better sorting.
589 HRTF data sets specified in the hrtf_tables config option may now be
590 relative or absolute filenames.
592 Made the default HRTF data set an external file, and added a data set for
593 48khz playback in addition to 44.1khz.
595 Added support for C11 atomic methods.
597 Improved support for some non-GNU build systems.
601 Fixed a regression with retrieving the source's AL_GAIN property.
605 Fixed device enumeration with the OSS backend.
607 Reorganized internal mixing logic, so unneeded steps can potentially be
608 skipped for better performance.
610 Removed the lookup table for calculating the mixing pans. The panning is
611 now calculated directly for better precision.
613 Improved the panning of stereo source channels when using stereo output.
615 Improved source filter quality on send paths.
617 Added a config option to allow PulseAudio to move streams between devices.
619 The PulseAudio backend will now attempt to spawn a server by default.
621 Added a workaround for a DirectSound bug relating to float32 output.
623 Added SSE-based mixers, for HRTF and non-HRTF mixing.
625 Added support for the new AL_SOFT_source_latency extension.
627 Improved ALSA capture by avoiding an extra buffer when using sizes
628 supported by the underlying device.
630 Improved the makehrtf utility to support new options and input formats.
632 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
633 the header includes can optionally be omitted.
635 Added a couple example code programs to show how to apply reverb, and
638 The configuration sample is now installed into the share/openal/ directory
639 instead of /etc/openal.
641 The configuration sample now gets installed by default.