3 Implemented the ALC_SOFT_reopen_device extension. This allows for moving
4 devices to different outputs without losing object state.
6 Implemented the legacy EAX extensions. Enabled by default only on Windows.
8 Improved sound positioning stability when a source is near the listener.
10 Improved the default 5.1 output decoder.
12 Improved the high frequency response for the HRTF second-order ambisonic
15 Improved SoundIO capture behavior.
17 Fixed UHJ output on NEON-capable CPUs.
19 Fixed redundant effect updates when setting an effect property to the
22 Fixed WASAPI capture using really low sample rates, and sources with very
23 high pitch shifts when using a bsinc resampler.
25 Added a PipeWire backend.
27 Added enumeration for the JACK and CoreAudio backends.
29 Added optional support for RTKit to get real-time priority. Only used as a
30 backup when pthread_setschedparam fails.
32 Added an option for JACK playback to render directly in the real-time
33 processing callback. For lower playback latency, on by default.
35 Added an option for custom JACK devices.
37 Added utilities to encode and decode UHJ audio files. Files are decoded to
38 the .amb format, and are encoded from libsndfile-compatible formats.
40 Added an in-progress extension to handle UHJ audio buffers natively.
42 Added an in-progress extension to select UHJ output in addition to HRTF.
44 Added an in-progress extension to hold sources in a playing state when a
45 device disconnects. Allows devices to be reset or reopened and have sources
46 resume from where they left off.
48 Lowered the priority of the JACK backend. To avoid it getting picked when
49 PipeWire is providing JACK compatibility, since the JACK backend is less
50 robust with auto-configuration.
54 Improved alext.h's detection of standard types.
56 Improved slightly the local source position when the listener and source
59 Improved click/pop prevention for sounds that stop prematurely.
61 Fixed compilation for Windows ARM targets with MSVC.
63 Fixed ARM NEON detection on Windows.
65 Fixed CoreAudio capture when the requested sample rate doesn't match the
68 Fixed OpenSL capture desyncing from the internal capture buffer.
70 Fixed sources missing a batch update when applied after quickly restarting
73 Fixed missing source stop events when stopping a paused source.
75 Added capture support to the experimental Oboe backend.
79 Updated library codebase to C++14.
81 Implemented the AL_SOFT_effect_target extension.
83 Implemented the AL_SOFT_events extension.
85 Implemented the ALC_SOFT_loopback_bformat extension.
87 Improved memory use for mixing voices.
89 Improved detection of NEON capabilities.
91 Improved handling of PulseAudio devices that lack manual start control.
93 Improved mixing performance with PulseAudio.
95 Improved high-frequency scaling quality for the HRTF B-Format decoder.
97 Improved makemhr's HRIR delay calculation.
99 Improved WASAPI capture of mono formats with multichannel input.
101 Reimplemented the modulation stage for reverb.
103 Enabled real-time mixing priority by default, for backends that use the
104 setting. It can still be disabled in the config file.
106 Enabled dual-band processing for the built-in quad and 7.1 output decoders.
108 Fixed a potential crash when deleting an effect slot immediately after the
109 last source using it stops.
111 Fixed building with the static runtime on MSVC.
113 Fixed using source stereo angles outside of -pi...+pi.
115 Fixed the buffer processed event count for sources that start with empty
118 Fixed trying to open an unopenable WASAPI device causing all devices to
121 Fixed stale devices when re-enumerating WASAPI devices.
123 Fixed using unicode paths with the log file on Windows.
125 Fixed DirectSound capture reporting bad sample counts or erroring when
128 Added an in-progress extension for a callback-driven buffer type.
130 Added an in-progress extension for higher-order B-Format buffers.
132 Added an in-progress extension for convolution reverb.
134 Added an experimental Oboe backend for Android playback. This requires the
135 Oboe sources at build time, so that it's built as a static library included
138 Added an option for auto-connecting JACK ports.
140 Added greater-than-stereo support to the SoundIO backend.
142 Modified the mixer to be fully asynchronous with the external API, and
143 should now be real-time safe. Although alcRenderSamplesSOFT is not due to
144 locking to check the device handle validity.
146 Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
147 to non-filtered signal phase.
149 Converted examples from SDL_sound to libsndfile. To avoid issues when
150 combining SDL2 and SDL_sound.
152 Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
153 https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
155 Reduced the maximum number of source sends from 16 to 6.
157 Removed the QSA backend. It's been broken for who knows how long.
159 Got rid of the compile-time native-tools targets, using cmake and global
160 initialization instead. This should make cross-compiling less troublesome.
164 Implemented the AL_SOFT_direct_channels_remix extension. This extends
165 AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
166 a matching output channel.
168 Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
169 support for N3D or SN3D scaling, or ACN channel ordering.
171 Fixed a potential voice leak when a source is started and stopped or
172 restarted in quick succession.
174 Fixed a potential device reset failure with JACK.
176 Improved handling of unsupported channel configurations with WASAPI. Such
177 setups will now try to output at least a stereo mix.
179 Improved clarity a bit for the HRTF second-order ambisonic decoder.
181 Improved detection of compatible layouts for SOFA files in makemhr and
184 Added the ability to resample HRTFs on load. MHR files no longer need to
185 match the device sample rate to be usable.
187 Added an option to limit the HRTF's filter length.
191 Converted the library codebase to C++11. A lot of hacks and custom
192 structures have been replaced with standard or cleaner implementations.
194 Partially implemented the Vocal Morpher effect.
196 Fixed the bsinc SSE resamplers on non-GCC compilers.
198 Fixed OpenSL capture.
200 Fixed support for extended capture formats with OpenSL.
202 Fixed handling of WASAPI not reporting a default device.
204 Fixed performance problems relating to semaphores on macOS.
206 Modified the bsinc12 resampler's transition band to better avoid aliasing
209 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
211 Modified the virtual speaker layout for HRTF B-Format decoding.
213 Modified the PulseAudio backend to use a custom processing loop.
215 Renamed the makehrtf utility to makemhr.
217 Improved the efficiency of the bsinc resamplers when up-sampling.
219 Improved the quality of the bsinc resamplers slightly.
221 Improved the efficiency of the HRTF filters.
223 Improved the HRTF B-Format decoder coefficient generation.
225 Improved reverb feedback fading to be more consistent with pan fading.
227 Improved handling of sources that end prematurely, avoiding loud clicks.
229 Improved the performance of some reverb processing loops.
231 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
232 some quality. Notably, down-sampling has less smooth pitch ramping.
234 Added support for SOFA input files with makemhr.
236 Added a build option to use pre-built native tools. For cross-compiling,
237 use with caution and ensure the native tools' binaries are kept up-to-date.
239 Added an adjust-latency config option for the PulseAudio backend.
241 Added basic support for multi-field HRTFs.
243 Added an option for mixing first- or second-order B-Format with HRTF
244 output. This can improve HRTF performance given a number of sources.
246 Added an RC file for proper DLL version information.
248 Disabled some old KDE workarounds by default. Specifically, PulseAudio
249 streams can now be moved (KDE may try to move them after opening).
253 Implemented capture support for the SoundIO backend.
255 Fixed source buffer queues potentially not playing properly when a queue
258 Fixed possible unexpected failures when generating auxiliary effect slots.
260 Fixed a crash with certain reverb or device settings.
262 Fixed OpenSL capture.
264 Improved output limiter response, better ensuring the sample amplitude is
269 Implemented the ALC_SOFT_device_clock extension.
271 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
273 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
275 Fixed compiling on NetBSD.
277 Fixed the reverb effect's density scale and panning parameters.
279 Fixed use of the WASAPI backend with certain games, which caused odd COM
280 initialization errors.
282 Increased the number of virtual channels for decoding Ambisonics to HRTF
285 Changed 32-bit x86 builds to use SSE2 math by default for performance.
286 Build-time options are available to use just SSE1 or x87 instead.
288 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
290 Renamed the MMDevAPI backend to WASAPI.
292 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
293 has been updated to 24-bit.
295 Added a 24- to 48-point band-limited Sinc resampler.
297 Added an SDL2 playback backend. Disabled by default to avoid a dependency
300 Improved the performance and quality of the Chorus and Flanger effects.
302 Improved the efficiency of the band-limited Sinc resampler.
304 Improved the Sinc resampler's transition band to avoid over-attenuating
307 Improved the performance of some filter operations.
309 Improved the efficiency of object ID lookups.
311 Improved the efficienty of internal voice/source synchronization.
313 Improved AL call error logging with contextualized messages.
315 Removed the reverb effect's modulation stage. Due to the lack of reference
316 for its intended behavior and strength.
320 Fixed resetting the FPU rounding mode after certain function calls on
323 Fixed use of SSE intrinsics when building with Clang on Windows.
325 Fixed a crash with the JACK backend when using JACK1.
327 Fixed use of pthread_setnane_np on NetBSD.
329 Fixed building on FreeBSD with an older freebsd-lib.
331 OSS now links with libossaudio if found at build time (for NetBSD).
335 Fixed an issue where resuming a source might not restart playing it.
337 Fixed PulseAudio playback when the configured stream length is much less
338 than the requested length.
340 Fixed MMDevAPI capture with sample rates not matching the backing device.
342 Fixed int32 output for the Wave Writer.
344 Fixed enumeration of OSS devices that are missing device files.
346 Added correct retrieval of the executable's path on FreeBSD.
348 Added a config option to specify the dithering depth.
350 Added a 5.1 decoder preset that excludes front-center output.
354 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
356 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
357 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
359 Implemented 3D processing for some effects. Currently implemented for
360 Reverb, Compressor, Equalizer, and Ring Modulator.
362 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
363 config option to be used.
365 Implemented dual-band processing for high-quality ambisonic decoding.
367 Implemented distance-compensation for surround sound output.
369 Implemented near-field emulation and compensation with ambisonic rendering.
370 Currently only applies when using the high-quality ambisonic decoder or
371 ambisonic output, with appropriate config options.
373 Implemented an output limiter to reduce the amount of distortion from
376 Implemented dithering for 8-bit and 16-bit output.
378 Implemented a config option to select a preferred HRTF.
380 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
382 Implemented experimental capture support for the OpenSL backend.
384 Fixed building on compilers with NEON support but don't default to having
387 Fixed support for JACK on Windows.
389 Fixed starting a source while alcSuspendContext is in effect.
391 Fixed detection of headsets as headphones, with MMDevAPI.
393 Added support for AmbDec config files, for custom ambisonic decoder
394 configurations. Version 3 files only.
396 Added backend-specific options to alsoft-config.
398 Added first-, second-, and third-order ambisonic output formats. Currently
399 only works with backends that don't rely on channel labels, like JACK,
402 Added a build option to embed the default HRTFs into the lib.
404 Added AmbDec presets to enable high-quality ambisonic decoding.
406 Added an AmbDec preset for 3D7.1 speaker setups.
408 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
409 the provided ambdec presets.
411 Added the ability for MMDevAPI to open devices given a Device ID or GUID
414 Added an option to the example apps to open a specific device.
416 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
417 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
420 Increased the default auxiliary effect slot count to 64 (up from 4).
422 Reduced the default period count to 3 (down from 4).
424 Slightly improved automatic naming for enumerated HRTFs.
426 Improved B-Format decoding with HRTF output.
428 Improved internal property handling for better batching behavior.
430 Improved performance of certain filter uses.
432 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
433 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
437 Implemented device enumeration for OSSv4.
439 Fixed building on OSX.
441 Fixed building on non-Windows systems without POSIX-2008.
443 Fixed Dedicated Dialog and Dedicated LFE effect output.
445 Added a build option to override the share install dir.
447 Added a build option to static-link libgcc for MinGW.
451 Fixed building with JACK and without PulseAudio.
453 Fixed building on FreeBSD.
455 Fixed the ALSA backend's allow-resampler option.
457 Fixed handling of inexact ALSA period counts.
459 Altered device naming scheme on Windows backends to better match other
462 Updated the CoreAudio backend to use the AudioComponent API. This clears up
463 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
467 Implemented a JACK playback backend.
469 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
471 Implemented the ALC_SOFT_HRTF extension.
473 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
475 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
476 24-point Sinc resampling, and performs anti-aliasing.
478 Implemented B-Format output support for the wave file writer. This creates
479 FuMa-style first-order Ambisonics wave files (AMB format).
481 Implemented a stereo-mode config option for treating stereo modes as either
482 speakers or headphones.
484 Implemented per-device configuration options.
486 Fixed handling of PulseAudio and MMDevAPI devices that have identical
489 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
491 Fixed logging of Unicode characters on Windows.
493 Fixed 5.1 surround sound channels. By default it will now use the side
494 channels for the surround output. A configuration using rear channels is
497 Fixed the QSA backend potentially altering the capture format.
499 Fixed detecting MMDevAPI's default device.
501 Fixed returning the default capture device name.
503 Fixed mixing property calculations when deferring context updates.
505 Altered the behavior of alcSuspendContext and alcProcessContext to better
506 match certain Windows drivers.
508 Altered the panning algorithm, utilizing Ambisonics for better side and
509 back positioning cues with surround sound output.
511 Improved support for certain older Windows apps.
513 Improved the alffplay example to support surround sound streams.
515 Improved support for building as a sub-project.
517 Added an HRTF playback example.
519 Added a tone generator output test.
521 Added a toolchain to help with cross-compiling to Android.
525 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
528 Implemented high-pass and band-pass EFX filters.
530 Implemented the high-pass filter for the EAXReverb effect.
532 Implemented SSE2 and SSE4.1 linear resamplers.
534 Implemented Neon-enhanced non-HRTF mixers.
536 Implemented a QSA backend, for QNX.
538 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
539 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
542 Fixed resetting mmdevapi backend devices.
544 Fixed clamping when converting 32-bit float samples to integer.
546 Fixed modulation range in the Modulator effect.
548 Several fixes for the OpenSL playback backend.
550 Fixed device specifier names that have Unicode characters on Windows.
552 Added support for filenames and paths with Unicode (UTF-8) characters on
555 Added support for alsoft.conf config files found in XDG Base Directory
556 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
557 defaults) on non-Windows systems.
559 Added a GUI configuration utility (requires Qt 4.8).
561 Added support for environment variable expansion in config options (not
562 keys or section names).
564 Added an example that uses SDL2 and ffmpeg.
566 Modified examples to use SDL_sound.
568 Modified CMake config option names for better sorting.
570 HRTF data sets specified in the hrtf_tables config option may now be
571 relative or absolute filenames.
573 Made the default HRTF data set an external file, and added a data set for
574 48khz playback in addition to 44.1khz.
576 Added support for C11 atomic methods.
578 Improved support for some non-GNU build systems.
582 Fixed a regression with retrieving the source's AL_GAIN property.
586 Fixed device enumeration with the OSS backend.
588 Reorganized internal mixing logic, so unneeded steps can potentially be
589 skipped for better performance.
591 Removed the lookup table for calculating the mixing pans. The panning is
592 now calculated directly for better precision.
594 Improved the panning of stereo source channels when using stereo output.
596 Improved source filter quality on send paths.
598 Added a config option to allow PulseAudio to move streams between devices.
600 The PulseAudio backend will now attempt to spawn a server by default.
602 Added a workaround for a DirectSound bug relating to float32 output.
604 Added SSE-based mixers, for HRTF and non-HRTF mixing.
606 Added support for the new AL_SOFT_source_latency extension.
608 Improved ALSA capture by avoiding an extra buffer when using sizes
609 supported by the underlying device.
611 Improved the makehrtf utility to support new options and input formats.
613 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
614 the header includes can optionally be omitted.
616 Added a couple example code programs to show how to apply reverb, and
619 The configuration sample is now installed into the share/openal/ directory
620 instead of /etc/openal.
622 The configuration sample now gets installed by default.