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