3 Updated library codebase to C++14.
5 Implemented the AL_SOFT_effect_target extension.
7 Implemented the AL_SOFT_events extension.
9 Implemented the ALC_SOFT_loopback_bformat extension.
11 Improved memory use for mixing voices.
13 Improved detection of NEON capabilities.
15 Improved handling of PulseAudio devices that lack manual start control.
17 Improved mixing performance with PulseAudio.
19 Improved high-frequency scaling quality for the HRTF B-Format decoder.
21 Improved makemhr's HRIR delay calculation.
23 Improved WASAPI capture of mono formats with multichannel input.
25 Reimplemented the modulation stage for reverb.
27 Enabled real-time mixing priority by default, for backends that use the
28 setting. It can still be disabled in the config file.
30 Enabled dual-band processing for the built-in quad and 7.1 output decoders.
32 Fixed a potential crash when deleting an effect slot immediately after the
33 last source using it stops.
35 Fixed building with the static runtime on MSVC.
37 Fixed using source stereo angles outside of -pi...+pi.
39 Fixed the buffer processed event count for sources that start with empty
42 Fixed trying to open an unopenable WASAPI device causing all devices to
45 Fixed stale devices when re-enumerating WASAPI devices.
47 Fixed using unicode paths with the log file on Windows.
49 Fixed DirectSound capture reporting bad sample counts or erroring when
52 Added an in-progress extension for a callback-driven buffer type.
54 Added an in-progress extension for higher-order B-Format buffers.
56 Added an in-progress extension for convolution reverb.
58 Added an experimental Oboe backend for Android playback. This requires the
59 Oboe sources at build time, so that it's built as a static library included
62 Added an option for auto-connecting JACK ports.
64 Added greater-than-stereo support to the SoundIO backend.
66 Modified the mixer to be fully asynchronous with the external API, and
67 should now be real-time safe. Although alcRenderSamplesSOFT is not due to
68 locking to check the device handle validity.
70 Modified the UHJ encoder to use an all-pass FIR filter that's less harmful
71 to non-filtered signal phase.
73 Converted examples from SDL_sound to libsndfile. To avoid issues when
74 combining SDL2 and SDL_sound.
76 Worked around a 32-bit GCC/MinGW bug with TLS destructors. See:
77 https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83562
79 Reduced the maximum number of source sends from 16 to 6.
81 Removed the QSA backend. It's been broken for who knows how long.
83 Got rid of the compile-time native-tools targets, using cmake and global
84 initialization instead. This should make cross-compiling less troublesome.
88 Implemented the AL_SOFT_direct_channels_remix extension. This extends
89 AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
90 a matching output channel.
92 Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
93 support for N3D or SN3D scaling, or ACN channel ordering.
95 Fixed a potential voice leak when a source is started and stopped or
96 restarted in quick succession.
98 Fixed a potential device reset failure with JACK.
100 Improved handling of unsupported channel configurations with WASAPI. Such
101 setups will now try to output at least a stereo mix.
103 Improved clarity a bit for the HRTF second-order ambisonic decoder.
105 Improved detection of compatible layouts for SOFA files in makemhr and
108 Added the ability to resample HRTFs on load. MHR files no longer need to
109 match the device sample rate to be usable.
111 Added an option to limit the HRTF's filter length.
115 Converted the library codebase to C++11. A lot of hacks and custom
116 structures have been replaced with standard or cleaner implementations.
118 Partially implemented the Vocal Morpher effect.
120 Fixed the bsinc SSE resamplers on non-GCC compilers.
122 Fixed OpenSL capture.
124 Fixed support for extended capture formats with OpenSL.
126 Fixed handling of WASAPI not reporting a default device.
128 Fixed performance problems relating to semaphores on macOS.
130 Modified the bsinc12 resampler's transition band to better avoid aliasing
133 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
135 Modified the virtual speaker layout for HRTF B-Format decoding.
137 Modified the PulseAudio backend to use a custom processing loop.
139 Renamed the makehrtf utility to makemhr.
141 Improved the efficiency of the bsinc resamplers when up-sampling.
143 Improved the quality of the bsinc resamplers slightly.
145 Improved the efficiency of the HRTF filters.
147 Improved the HRTF B-Format decoder coefficient generation.
149 Improved reverb feedback fading to be more consistent with pan fading.
151 Improved handling of sources that end prematurely, avoiding loud clicks.
153 Improved the performance of some reverb processing loops.
155 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
156 some quality. Notably, down-sampling has less smooth pitch ramping.
158 Added support for SOFA input files with makemhr.
160 Added a build option to use pre-built native tools. For cross-compiling,
161 use with caution and ensure the native tools' binaries are kept up-to-date.
163 Added an adjust-latency config option for the PulseAudio backend.
165 Added basic support for multi-field HRTFs.
167 Added an option for mixing first- or second-order B-Format with HRTF
168 output. This can improve HRTF performance given a number of sources.
170 Added an RC file for proper DLL version information.
172 Disabled some old KDE workarounds by default. Specifically, PulseAudio
173 streams can now be moved (KDE may try to move them after opening).
177 Implemented capture support for the SoundIO backend.
179 Fixed source buffer queues potentially not playing properly when a queue
182 Fixed possible unexpected failures when generating auxiliary effect slots.
184 Fixed a crash with certain reverb or device settings.
186 Fixed OpenSL capture.
188 Improved output limiter response, better ensuring the sample amplitude is
193 Implemented the ALC_SOFT_device_clock extension.
195 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
197 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
199 Fixed compiling on NetBSD.
201 Fixed the reverb effect's density scale and panning parameters.
203 Fixed use of the WASAPI backend with certain games, which caused odd COM
204 initialization errors.
206 Increased the number of virtual channels for decoding Ambisonics to HRTF
209 Changed 32-bit x86 builds to use SSE2 math by default for performance.
210 Build-time options are available to use just SSE1 or x87 instead.
212 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
214 Renamed the MMDevAPI backend to WASAPI.
216 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
217 has been updated to 24-bit.
219 Added a 24- to 48-point band-limited Sinc resampler.
221 Added an SDL2 playback backend. Disabled by default to avoid a dependency
224 Improved the performance and quality of the Chorus and Flanger effects.
226 Improved the efficiency of the band-limited Sinc resampler.
228 Improved the Sinc resampler's transition band to avoid over-attenuating
231 Improved the performance of some filter operations.
233 Improved the efficiency of object ID lookups.
235 Improved the efficienty of internal voice/source synchronization.
237 Improved AL call error logging with contextualized messages.
239 Removed the reverb effect's modulation stage. Due to the lack of reference
240 for its intended behavior and strength.
244 Fixed resetting the FPU rounding mode after certain function calls on
247 Fixed use of SSE intrinsics when building with Clang on Windows.
249 Fixed a crash with the JACK backend when using JACK1.
251 Fixed use of pthread_setnane_np on NetBSD.
253 Fixed building on FreeBSD with an older freebsd-lib.
255 OSS now links with libossaudio if found at build time (for NetBSD).
259 Fixed an issue where resuming a source might not restart playing it.
261 Fixed PulseAudio playback when the configured stream length is much less
262 than the requested length.
264 Fixed MMDevAPI capture with sample rates not matching the backing device.
266 Fixed int32 output for the Wave Writer.
268 Fixed enumeration of OSS devices that are missing device files.
270 Added correct retrieval of the executable's path on FreeBSD.
272 Added a config option to specify the dithering depth.
274 Added a 5.1 decoder preset that excludes front-center output.
278 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
280 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
281 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
283 Implemented 3D processing for some effects. Currently implemented for
284 Reverb, Compressor, Equalizer, and Ring Modulator.
286 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
287 config option to be used.
289 Implemented dual-band processing for high-quality ambisonic decoding.
291 Implemented distance-compensation for surround sound output.
293 Implemented near-field emulation and compensation with ambisonic rendering.
294 Currently only applies when using the high-quality ambisonic decoder or
295 ambisonic output, with appropriate config options.
297 Implemented an output limiter to reduce the amount of distortion from
300 Implemented dithering for 8-bit and 16-bit output.
302 Implemented a config option to select a preferred HRTF.
304 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
306 Implemented experimental capture support for the OpenSL backend.
308 Fixed building on compilers with NEON support but don't default to having
311 Fixed support for JACK on Windows.
313 Fixed starting a source while alcSuspendContext is in effect.
315 Fixed detection of headsets as headphones, with MMDevAPI.
317 Added support for AmbDec config files, for custom ambisonic decoder
318 configurations. Version 3 files only.
320 Added backend-specific options to alsoft-config.
322 Added first-, second-, and third-order ambisonic output formats. Currently
323 only works with backends that don't rely on channel labels, like JACK,
326 Added a build option to embed the default HRTFs into the lib.
328 Added AmbDec presets to enable high-quality ambisonic decoding.
330 Added an AmbDec preset for 3D7.1 speaker setups.
332 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
333 the provided ambdec presets.
335 Added the ability for MMDevAPI to open devices given a Device ID or GUID
338 Added an option to the example apps to open a specific device.
340 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
341 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
344 Increased the default auxiliary effect slot count to 64 (up from 4).
346 Reduced the default period count to 3 (down from 4).
348 Slightly improved automatic naming for enumerated HRTFs.
350 Improved B-Format decoding with HRTF output.
352 Improved internal property handling for better batching behavior.
354 Improved performance of certain filter uses.
356 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
357 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
361 Implemented device enumeration for OSSv4.
363 Fixed building on OSX.
365 Fixed building on non-Windows systems without POSIX-2008.
367 Fixed Dedicated Dialog and Dedicated LFE effect output.
369 Added a build option to override the share install dir.
371 Added a build option to static-link libgcc for MinGW.
375 Fixed building with JACK and without PulseAudio.
377 Fixed building on FreeBSD.
379 Fixed the ALSA backend's allow-resampler option.
381 Fixed handling of inexact ALSA period counts.
383 Altered device naming scheme on Windows backends to better match other
386 Updated the CoreAudio backend to use the AudioComponent API. This clears up
387 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
391 Implemented a JACK playback backend.
393 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
395 Implemented the ALC_SOFT_HRTF extension.
397 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
399 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
400 24-point Sinc resampling, and performs anti-aliasing.
402 Implemented B-Format output support for the wave file writer. This creates
403 FuMa-style first-order Ambisonics wave files (AMB format).
405 Implemented a stereo-mode config option for treating stereo modes as either
406 speakers or headphones.
408 Implemented per-device configuration options.
410 Fixed handling of PulseAudio and MMDevAPI devices that have identical
413 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
415 Fixed logging of Unicode characters on Windows.
417 Fixed 5.1 surround sound channels. By default it will now use the side
418 channels for the surround output. A configuration using rear channels is
421 Fixed the QSA backend potentially altering the capture format.
423 Fixed detecting MMDevAPI's default device.
425 Fixed returning the default capture device name.
427 Fixed mixing property calculations when deferring context updates.
429 Altered the behavior of alcSuspendContext and alcProcessContext to better
430 match certain Windows drivers.
432 Altered the panning algorithm, utilizing Ambisonics for better side and
433 back positioning cues with surround sound output.
435 Improved support for certain older Windows apps.
437 Improved the alffplay example to support surround sound streams.
439 Improved support for building as a sub-project.
441 Added an HRTF playback example.
443 Added a tone generator output test.
445 Added a toolchain to help with cross-compiling to Android.
449 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
452 Implemented high-pass and band-pass EFX filters.
454 Implemented the high-pass filter for the EAXReverb effect.
456 Implemented SSE2 and SSE4.1 linear resamplers.
458 Implemented Neon-enhanced non-HRTF mixers.
460 Implemented a QSA backend, for QNX.
462 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
463 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
466 Fixed resetting mmdevapi backend devices.
468 Fixed clamping when converting 32-bit float samples to integer.
470 Fixed modulation range in the Modulator effect.
472 Several fixes for the OpenSL playback backend.
474 Fixed device specifier names that have Unicode characters on Windows.
476 Added support for filenames and paths with Unicode (UTF-8) characters on
479 Added support for alsoft.conf config files found in XDG Base Directory
480 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
481 defaults) on non-Windows systems.
483 Added a GUI configuration utility (requires Qt 4.8).
485 Added support for environment variable expansion in config options (not
486 keys or section names).
488 Added an example that uses SDL2 and ffmpeg.
490 Modified examples to use SDL_sound.
492 Modified CMake config option names for better sorting.
494 HRTF data sets specified in the hrtf_tables config option may now be
495 relative or absolute filenames.
497 Made the default HRTF data set an external file, and added a data set for
498 48khz playback in addition to 44.1khz.
500 Added support for C11 atomic methods.
502 Improved support for some non-GNU build systems.
506 Fixed a regression with retrieving the source's AL_GAIN property.
510 Fixed device enumeration with the OSS backend.
512 Reorganized internal mixing logic, so unneeded steps can potentially be
513 skipped for better performance.
515 Removed the lookup table for calculating the mixing pans. The panning is
516 now calculated directly for better precision.
518 Improved the panning of stereo source channels when using stereo output.
520 Improved source filter quality on send paths.
522 Added a config option to allow PulseAudio to move streams between devices.
524 The PulseAudio backend will now attempt to spawn a server by default.
526 Added a workaround for a DirectSound bug relating to float32 output.
528 Added SSE-based mixers, for HRTF and non-HRTF mixing.
530 Added support for the new AL_SOFT_source_latency extension.
532 Improved ALSA capture by avoiding an extra buffer when using sizes
533 supported by the underlying device.
535 Improved the makehrtf utility to support new options and input formats.
537 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
538 the header includes can optionally be omitted.
540 Added a couple example code programs to show how to apply reverb, and
543 The configuration sample is now installed into the share/openal/ directory
544 instead of /etc/openal.
546 The configuration sample now gets installed by default.