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