3 Updated library codebase to C++14.
5 Improved memory use for mixing voices.
7 Improved detection of NEON capabilities.
9 Improved handling of PulseAudio devices that lack manual start control.
11 Improved mixing performance with PulseAudio.
13 Improved high-frequency scaling quality for the HRTF B-Format decoder.
15 Improved makemhr's HRIR delay calculation.
17 Reimplemented the modulation stage for reverb.
19 Enabled real-time mixing priority by default, for backends that use the
20 setting. It can still be disabled in the config file.
22 Enabled dual-band processing for the built-in quad and 7.1 output decoders.
24 Fixed building with the static runtime on MSVC.
26 Fixed using source stereo angles outside of -pi...+pi.
28 Fixed the buffer processed event count for sources that start with empty
31 Fixed trying to open an unopenable WASAPI device causing all devices to
34 Fixed stale devices when re-enumerating WASAPI devices.
36 Fixed using unicode paths with the log file on Windows.
38 Added an in-progress extension for a callback-driven buffer type.
40 Added an in-progress extension for higher-order B-Format buffers.
42 Added an experimental Oboe backend for Android playback. This requires the
43 Oboe sources at build time, so that it's built as a static library included
46 Added an option for auto-connecting JACK ports.
48 Added greater-than-stereo support to the SoundIO backend.
50 Modified the mixer to be fully asynchronous with the external API, and
51 should now be real-time safe. Although alcRenderSamplesSOFT is not due to
52 locking to check the device handle validity.
54 Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
55 to non-filtered signal phase.
57 Converted examples from SDL_sound to libsndfile. To avoid issues when
58 combining SDL2 and SDL_sound.
60 Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
61 https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
63 Reduced the maximum number of source sends from 16 to 6.
65 Removed the QSA backend. It's been broken for who knows how long.
67 Got rid of the compile-time native-tools targets, using cmake and global
68 initialization instead. This should make cross-compiling less troublesome.
74 Implemented the AL_SOFT_direct_channels_remix extension. This extends
75 AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
76 a matching output channel.
78 Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
79 support for N3D or SN3D scaling, or ACN channel ordering.
81 Fixed a potential voice leak when a source is started and stopped or
82 restarted in quick succession.
84 Fixed a potential device reset failure with JACK.
86 Improved handling of unsupported channel configurations with WASAPI. Such
87 setups will now try to output at least a stereo mix.
89 Improved clarity a bit for the HRTF second-order ambisonic decoder.
91 Improved detection of compatible layouts for SOFA files in makemhr and
94 Added the ability to resample HRTFs on load. MHR files no longer need to
95 match the device sample rate to be usable.
97 Added an option to limit the HRTF's filter length.
101 Converted the library codebase to C++11. A lot of hacks and custom
102 structures have been replaced with standard or cleaner implementations.
104 Partially implemented the Vocal Morpher effect.
106 Fixed the bsinc SSE resamplers on non-GCC compilers.
108 Fixed OpenSL capture.
110 Fixed support for extended capture formats with OpenSL.
112 Fixed handling of WASAPI not reporting a default device.
114 Fixed performance problems relating to semaphores on macOS.
116 Modified the bsinc12 resampler's transition band to better avoid aliasing
119 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
121 Modified the virtual speaker layout for HRTF B-Format decoding.
123 Modified the PulseAudio backend to use a custom processing loop.
125 Renamed the makehrtf utility to makemhr.
127 Improved the efficiency of the bsinc resamplers when up-sampling.
129 Improved the quality of the bsinc resamplers slightly.
131 Improved the efficiency of the HRTF filters.
133 Improved the HRTF B-Format decoder coefficient generation.
135 Improved reverb feedback fading to be more consistent with pan fading.
137 Improved handling of sources that end prematurely, avoiding loud clicks.
139 Improved the performance of some reverb processing loops.
141 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
142 some quality. Notably, down-sampling has less smooth pitch ramping.
144 Added support for SOFA input files with makemhr.
146 Added a build option to use pre-built native tools. For cross-compiling,
147 use with caution and ensure the native tools' binaries are kept up-to-date.
149 Added an adjust-latency config option for the PulseAudio backend.
151 Added basic support for multi-field HRTFs.
153 Added an option for mixing first- or second-order B-Format with HRTF
154 output. This can improve HRTF performance given a number of sources.
156 Added an RC file for proper DLL version information.
158 Disabled some old KDE workarounds by default. Specifically, PulseAudio
159 streams can now be moved (KDE may try to move them after opening).
163 Implemented capture support for the SoundIO backend.
165 Fixed source buffer queues potentially not playing properly when a queue
168 Fixed possible unexpected failures when generating auxiliary effect slots.
170 Fixed a crash with certain reverb or device settings.
172 Fixed OpenSL capture.
174 Improved output limiter response, better ensuring the sample amplitude is
179 Implemented the ALC_SOFT_device_clock extension.
181 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
183 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
185 Fixed compiling on NetBSD.
187 Fixed the reverb effect's density scale and panning parameters.
189 Fixed use of the WASAPI backend with certain games, which caused odd COM
190 initialization errors.
192 Increased the number of virtual channels for decoding Ambisonics to HRTF
195 Changed 32-bit x86 builds to use SSE2 math by default for performance.
196 Build-time options are available to use just SSE1 or x87 instead.
198 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
200 Renamed the MMDevAPI backend to WASAPI.
202 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
203 has been updated to 24-bit.
205 Added a 24- to 48-point band-limited Sinc resampler.
207 Added an SDL2 playback backend. Disabled by default to avoid a dependency
210 Improved the performance and quality of the Chorus and Flanger effects.
212 Improved the efficiency of the band-limited Sinc resampler.
214 Improved the Sinc resampler's transition band to avoid over-attenuating
217 Improved the performance of some filter operations.
219 Improved the efficiency of object ID lookups.
221 Improved the efficienty of internal voice/source synchronization.
223 Improved AL call error logging with contextualized messages.
225 Removed the reverb effect's modulation stage. Due to the lack of reference
226 for its intended behavior and strength.
230 Fixed resetting the FPU rounding mode after certain function calls on
233 Fixed use of SSE intrinsics when building with Clang on Windows.
235 Fixed a crash with the JACK backend when using JACK1.
237 Fixed use of pthread_setnane_np on NetBSD.
239 Fixed building on FreeBSD with an older freebsd-lib.
241 OSS now links with libossaudio if found at build time (for NetBSD).
245 Fixed an issue where resuming a source might not restart playing it.
247 Fixed PulseAudio playback when the configured stream length is much less
248 than the requested length.
250 Fixed MMDevAPI capture with sample rates not matching the backing device.
252 Fixed int32 output for the Wave Writer.
254 Fixed enumeration of OSS devices that are missing device files.
256 Added correct retrieval of the executable's path on FreeBSD.
258 Added a config option to specify the dithering depth.
260 Added a 5.1 decoder preset that excludes front-center output.
264 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
266 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
267 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
269 Implemented 3D processing for some effects. Currently implemented for
270 Reverb, Compressor, Equalizer, and Ring Modulator.
272 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
273 config option to be used.
275 Implemented dual-band processing for high-quality ambisonic decoding.
277 Implemented distance-compensation for surround sound output.
279 Implemented near-field emulation and compensation with ambisonic rendering.
280 Currently only applies when using the high-quality ambisonic decoder or
281 ambisonic output, with appropriate config options.
283 Implemented an output limiter to reduce the amount of distortion from
286 Implemented dithering for 8-bit and 16-bit output.
288 Implemented a config option to select a preferred HRTF.
290 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
292 Implemented experimental capture support for the OpenSL backend.
294 Fixed building on compilers with NEON support but don't default to having
297 Fixed support for JACK on Windows.
299 Fixed starting a source while alcSuspendContext is in effect.
301 Fixed detection of headsets as headphones, with MMDevAPI.
303 Added support for AmbDec config files, for custom ambisonic decoder
304 configurations. Version 3 files only.
306 Added backend-specific options to alsoft-config.
308 Added first-, second-, and third-order ambisonic output formats. Currently
309 only works with backends that don't rely on channel labels, like JACK,
312 Added a build option to embed the default HRTFs into the lib.
314 Added AmbDec presets to enable high-quality ambisonic decoding.
316 Added an AmbDec preset for 3D7.1 speaker setups.
318 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
319 the provided ambdec presets.
321 Added the ability for MMDevAPI to open devices given a Device ID or GUID
324 Added an option to the example apps to open a specific device.
326 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
327 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
330 Increased the default auxiliary effect slot count to 64 (up from 4).
332 Reduced the default period count to 3 (down from 4).
334 Slightly improved automatic naming for enumerated HRTFs.
336 Improved B-Format decoding with HRTF output.
338 Improved internal property handling for better batching behavior.
340 Improved performance of certain filter uses.
342 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
343 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
347 Implemented device enumeration for OSSv4.
349 Fixed building on OSX.
351 Fixed building on non-Windows systems without POSIX-2008.
353 Fixed Dedicated Dialog and Dedicated LFE effect output.
355 Added a build option to override the share install dir.
357 Added a build option to static-link libgcc for MinGW.
361 Fixed building with JACK and without PulseAudio.
363 Fixed building on FreeBSD.
365 Fixed the ALSA backend's allow-resampler option.
367 Fixed handling of inexact ALSA period counts.
369 Altered device naming scheme on Windows backends to better match other
372 Updated the CoreAudio backend to use the AudioComponent API. This clears up
373 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
377 Implemented a JACK playback backend.
379 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
381 Implemented the ALC_SOFT_HRTF extension.
383 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
385 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
386 24-point Sinc resampling, and performs anti-aliasing.
388 Implemented B-Format output support for the wave file writer. This creates
389 FuMa-style first-order Ambisonics wave files (AMB format).
391 Implemented a stereo-mode config option for treating stereo modes as either
392 speakers or headphones.
394 Implemented per-device configuration options.
396 Fixed handling of PulseAudio and MMDevAPI devices that have identical
399 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
401 Fixed logging of Unicode characters on Windows.
403 Fixed 5.1 surround sound channels. By default it will now use the side
404 channels for the surround output. A configuration using rear channels is
407 Fixed the QSA backend potentially altering the capture format.
409 Fixed detecting MMDevAPI's default device.
411 Fixed returning the default capture device name.
413 Fixed mixing property calculations when deferring context updates.
415 Altered the behavior of alcSuspendContext and alcProcessContext to better
416 match certain Windows drivers.
418 Altered the panning algorithm, utilizing Ambisonics for better side and
419 back positioning cues with surround sound output.
421 Improved support for certain older Windows apps.
423 Improved the alffplay example to support surround sound streams.
425 Improved support for building as a sub-project.
427 Added an HRTF playback example.
429 Added a tone generator output test.
431 Added a toolchain to help with cross-compiling to Android.
435 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
438 Implemented high-pass and band-pass EFX filters.
440 Implemented the high-pass filter for the EAXReverb effect.
442 Implemented SSE2 and SSE4.1 linear resamplers.
444 Implemented Neon-enhanced non-HRTF mixers.
446 Implemented a QSA backend, for QNX.
448 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
449 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
452 Fixed resetting mmdevapi backend devices.
454 Fixed clamping when converting 32-bit float samples to integer.
456 Fixed modulation range in the Modulator effect.
458 Several fixes for the OpenSL playback backend.
460 Fixed device specifier names that have Unicode characters on Windows.
462 Added support for filenames and paths with Unicode (UTF-8) characters on
465 Added support for alsoft.conf config files found in XDG Base Directory
466 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
467 defaults) on non-Windows systems.
469 Added a GUI configuration utility (requires Qt 4.8).
471 Added support for environment variable expansion in config options (not
472 keys or section names).
474 Added an example that uses SDL2 and ffmpeg.
476 Modified examples to use SDL_sound.
478 Modified CMake config option names for better sorting.
480 HRTF data sets specified in the hrtf_tables config option may now be
481 relative or absolute filenames.
483 Made the default HRTF data set an external file, and added a data set for
484 48khz playback in addition to 44.1khz.
486 Added support for C11 atomic methods.
488 Improved support for some non-GNU build systems.
492 Fixed a regression with retrieving the source's AL_GAIN property.
496 Fixed device enumeration with the OSS backend.
498 Reorganized internal mixing logic, so unneeded steps can potentially be
499 skipped for better performance.
501 Removed the lookup table for calculating the mixing pans. The panning is
502 now calculated directly for better precision.
504 Improved the panning of stereo source channels when using stereo output.
506 Improved source filter quality on send paths.
508 Added a config option to allow PulseAudio to move streams between devices.
510 The PulseAudio backend will now attempt to spawn a server by default.
512 Added a workaround for a DirectSound bug relating to float32 output.
514 Added SSE-based mixers, for HRTF and non-HRTF mixing.
516 Added support for the new AL_SOFT_source_latency extension.
518 Improved ALSA capture by avoiding an extra buffer when using sizes
519 supported by the underlying device.
521 Improved the makehrtf utility to support new options and input formats.
523 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
524 the header includes can optionally be omitted.
526 Added a couple example code programs to show how to apply reverb, and
529 The configuration sample is now installed into the share/openal/ directory
530 instead of /etc/openal.
532 The configuration sample now gets installed by default.