3 Implemented the AL_SOFT_direct_channels_remix extension. This extends
4 AL_DIRECT_CHANNELS_SOFT to optionally remix input channels that don't have
5 a matching output channel.
7 Implemented the AL_SOFT_bformat_ex extension. This extends B-Format buffer
8 support for N3D or SN3D scaling, or ACN channel ordering.
10 Fixed a potential voice leak when a source is started and stopped or
11 restarted in quick succession.
13 Fixed a potential device reset failure with JACK.
15 Improved handling of unsupported channel configurations with WASAPI. Such
16 setups will now try to output at least a stereo mix.
18 Improved clarity a bit for the HRTF second-order ambisonic decoder.
20 Improved detection of compatible layouts for SOFA files in makemhr and
23 Added the ability to resample HRTFs on load. MHR files no longer need to
24 match the device sample rate to be usable.
26 Added an option to limit the HRTF's filter length.
30 Converted the library codebase to C++11. A lot of hacks and custom
31 structures have been replaced with standard or cleaner implementations.
33 Partially implemented the Vocal Morpher effect.
35 Fixed the bsinc SSE resamplers on non-GCC compilers.
39 Fixed support for extended capture formats with OpenSL.
41 Fixed handling of WASAPI not reporting a default device.
43 Fixed performance problems relating to semaphores on macOS.
45 Modified the bsinc12 resampler's transition band to better avoid aliasing
48 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
50 Modified the virtual speaker layout for HRTF B-Format decoding.
52 Modified the PulseAudio backend to use a custom processing loop.
54 Renamed the makehrtf utility to makemhr.
56 Improved the efficiency of the bsinc resamplers when up-sampling.
58 Improved the quality of the bsinc resamplers slightly.
60 Improved the efficiency of the HRTF filters.
62 Improved the HRTF B-Format decoder coefficient generation.
64 Improved reverb feedback fading to be more consistent with pan fading.
66 Improved handling of sources that end prematurely, avoiding loud clicks.
68 Improved the performance of some reverb processing loops.
70 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
71 some quality. Notably, down-sampling has less smooth pitch ramping.
73 Added support for SOFA input files with makemhr.
75 Added a build option to use pre-built native tools. For cross-compiling,
76 use with caution and ensure the native tools' binaries are kept up-to-date.
78 Added an adjust-latency config option for the PulseAudio backend.
80 Added basic support for multi-field HRTFs.
82 Added an option for mixing first- or second-order B-Format with HRTF
83 output. This can improve HRTF performance given a number of sources.
85 Added an RC file for proper DLL version information.
87 Disabled some old KDE workarounds by default. Specifically, PulseAudio
88 streams can now be moved (KDE may try to move them after opening).
92 Implemented capture support for the SoundIO backend.
94 Fixed source buffer queues potentially not playing properly when a queue
97 Fixed possible unexpected failures when generating auxiliary effect slots.
99 Fixed a crash with certain reverb or device settings.
101 Fixed OpenSL capture.
103 Improved output limiter response, better ensuring the sample amplitude is
108 Implemented the ALC_SOFT_device_clock extension.
110 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
112 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
114 Fixed compiling on NetBSD.
116 Fixed the reverb effect's density scale and panning parameters.
118 Fixed use of the WASAPI backend with certain games, which caused odd COM
119 initialization errors.
121 Increased the number of virtual channels for decoding Ambisonics to HRTF
124 Changed 32-bit x86 builds to use SSE2 math by default for performance.
125 Build-time options are available to use just SSE1 or x87 instead.
127 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
129 Renamed the MMDevAPI backend to WASAPI.
131 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
132 has been updated to 24-bit.
134 Added a 24- to 48-point band-limited Sinc resampler.
136 Added an SDL2 playback backend. Disabled by default to avoid a dependency
139 Improved the performance and quality of the Chorus and Flanger effects.
141 Improved the efficiency of the band-limited Sinc resampler.
143 Improved the Sinc resampler's transition band to avoid over-attenuating
146 Improved the performance of some filter operations.
148 Improved the efficiency of object ID lookups.
150 Improved the efficienty of internal voice/source synchronization.
152 Improved AL call error logging with contextualized messages.
154 Removed the reverb effect's modulation stage. Due to the lack of reference
155 for its intended behavior and strength.
159 Fixed resetting the FPU rounding mode after certain function calls on
162 Fixed use of SSE intrinsics when building with Clang on Windows.
164 Fixed a crash with the JACK backend when using JACK1.
166 Fixed use of pthread_setnane_np on NetBSD.
168 Fixed building on FreeBSD with an older freebsd-lib.
170 OSS now links with libossaudio if found at build time (for NetBSD).
174 Fixed an issue where resuming a source might not restart playing it.
176 Fixed PulseAudio playback when the configured stream length is much less
177 than the requested length.
179 Fixed MMDevAPI capture with sample rates not matching the backing device.
181 Fixed int32 output for the Wave Writer.
183 Fixed enumeration of OSS devices that are missing device files.
185 Added correct retrieval of the executable's path on FreeBSD.
187 Added a config option to specify the dithering depth.
189 Added a 5.1 decoder preset that excludes front-center output.
193 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
195 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
196 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
198 Implemented 3D processing for some effects. Currently implemented for
199 Reverb, Compressor, Equalizer, and Ring Modulator.
201 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
202 config option to be used.
204 Implemented dual-band processing for high-quality ambisonic decoding.
206 Implemented distance-compensation for surround sound output.
208 Implemented near-field emulation and compensation with ambisonic rendering.
209 Currently only applies when using the high-quality ambisonic decoder or
210 ambisonic output, with appropriate config options.
212 Implemented an output limiter to reduce the amount of distortion from
215 Implemented dithering for 8-bit and 16-bit output.
217 Implemented a config option to select a preferred HRTF.
219 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
221 Implemented experimental capture support for the OpenSL backend.
223 Fixed building on compilers with NEON support but don't default to having
226 Fixed support for JACK on Windows.
228 Fixed starting a source while alcSuspendContext is in effect.
230 Fixed detection of headsets as headphones, with MMDevAPI.
232 Added support for AmbDec config files, for custom ambisonic decoder
233 configurations. Version 3 files only.
235 Added backend-specific options to alsoft-config.
237 Added first-, second-, and third-order ambisonic output formats. Currently
238 only works with backends that don't rely on channel labels, like JACK,
241 Added a build option to embed the default HRTFs into the lib.
243 Added AmbDec presets to enable high-quality ambisonic decoding.
245 Added an AmbDec preset for 3D7.1 speaker setups.
247 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
248 the provided ambdec presets.
250 Added the ability for MMDevAPI to open devices given a Device ID or GUID
253 Added an option to the example apps to open a specific device.
255 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
256 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
259 Increased the default auxiliary effect slot count to 64 (up from 4).
261 Reduced the default period count to 3 (down from 4).
263 Slightly improved automatic naming for enumerated HRTFs.
265 Improved B-Format decoding with HRTF output.
267 Improved internal property handling for better batching behavior.
269 Improved performance of certain filter uses.
271 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
272 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
276 Implemented device enumeration for OSSv4.
278 Fixed building on OSX.
280 Fixed building on non-Windows systems without POSIX-2008.
282 Fixed Dedicated Dialog and Dedicated LFE effect output.
284 Added a build option to override the share install dir.
286 Added a build option to static-link libgcc for MinGW.
290 Fixed building with JACK and without PulseAudio.
292 Fixed building on FreeBSD.
294 Fixed the ALSA backend's allow-resampler option.
296 Fixed handling of inexact ALSA period counts.
298 Altered device naming scheme on Windows backends to better match other
301 Updated the CoreAudio backend to use the AudioComponent API. This clears up
302 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
306 Implemented a JACK playback backend.
308 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
310 Implemented the ALC_SOFT_HRTF extension.
312 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
314 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
315 24-point Sinc resampling, and performs anti-aliasing.
317 Implemented B-Format output support for the wave file writer. This creates
318 FuMa-style first-order Ambisonics wave files (AMB format).
320 Implemented a stereo-mode config option for treating stereo modes as either
321 speakers or headphones.
323 Implemented per-device configuration options.
325 Fixed handling of PulseAudio and MMDevAPI devices that have identical
328 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
330 Fixed logging of Unicode characters on Windows.
332 Fixed 5.1 surround sound channels. By default it will now use the side
333 channels for the surround output. A configuration using rear channels is
336 Fixed the QSA backend potentially altering the capture format.
338 Fixed detecting MMDevAPI's default device.
340 Fixed returning the default capture device name.
342 Fixed mixing property calculations when deferring context updates.
344 Altered the behavior of alcSuspendContext and alcProcessContext to better
345 match certain Windows drivers.
347 Altered the panning algorithm, utilizing Ambisonics for better side and
348 back positioning cues with surround sound output.
350 Improved support for certain older Windows apps.
352 Improved the alffplay example to support surround sound streams.
354 Improved support for building as a sub-project.
356 Added an HRTF playback example.
358 Added a tone generator output test.
360 Added a toolchain to help with cross-compiling to Android.
364 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
367 Implemented high-pass and band-pass EFX filters.
369 Implemented the high-pass filter for the EAXReverb effect.
371 Implemented SSE2 and SSE4.1 linear resamplers.
373 Implemented Neon-enhanced non-HRTF mixers.
375 Implemented a QSA backend, for QNX.
377 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
378 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
381 Fixed resetting mmdevapi backend devices.
383 Fixed clamping when converting 32-bit float samples to integer.
385 Fixed modulation range in the Modulator effect.
387 Several fixes for the OpenSL playback backend.
389 Fixed device specifier names that have Unicode characters on Windows.
391 Added support for filenames and paths with Unicode (UTF-8) characters on
394 Added support for alsoft.conf config files found in XDG Base Directory
395 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
396 defaults) on non-Windows systems.
398 Added a GUI configuration utility (requires Qt 4.8).
400 Added support for environment variable expansion in config options (not
401 keys or section names).
403 Added an example that uses SDL2 and ffmpeg.
405 Modified examples to use SDL_sound.
407 Modified CMake config option names for better sorting.
409 HRTF data sets specified in the hrtf_tables config option may now be
410 relative or absolute filenames.
412 Made the default HRTF data set an external file, and added a data set for
413 48khz playback in addition to 44.1khz.
415 Added support for C11 atomic methods.
417 Improved support for some non-GNU build systems.
421 Fixed a regression with retrieving the source's AL_GAIN property.
425 Fixed device enumeration with the OSS backend.
427 Reorganized internal mixing logic, so unneeded steps can potentially be
428 skipped for better performance.
430 Removed the lookup table for calculating the mixing pans. The panning is
431 now calculated directly for better precision.
433 Improved the panning of stereo source channels when using stereo output.
435 Improved source filter quality on send paths.
437 Added a config option to allow PulseAudio to move streams between devices.
439 The PulseAudio backend will now attempt to spawn a server by default.
441 Added a workaround for a DirectSound bug relating to float32 output.
443 Added SSE-based mixers, for HRTF and non-HRTF mixing.
445 Added support for the new AL_SOFT_source_latency extension.
447 Improved ALSA capture by avoiding an extra buffer when using sizes
448 supported by the underlying device.
450 Improved the makehrtf utility to support new options and input formats.
452 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
453 the header includes can optionally be omitted.
455 Added a couple example code programs to show how to apply reverb, and
458 The configuration sample is now installed into the share/openal/ directory
459 instead of /etc/openal.
461 The configuration sample now gets installed by default.