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