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 potential voice leak when a source is started and stopped in quick
13 Improved handling of unsupported channel configurations with WASAPI. Such
14 setups will now try to output at least a stereo mix.
16 Improved clarity a bit for the HRTF second-order ambisonic decoder.
18 Improved detection of compatible layouts for SOFA files in makemhr and
21 Added the ability to resample HRTFs on load. MHR files no longer need to
22 match the device sample rate to be usable.
24 Added an option to limit the HRTF's filter length.
28 Converted the library codebase to C++11. A lot of hacks and custom
29 structures have been replaced with standard or cleaner implementations.
31 Partially implemented the Vocal Morpher effect.
33 Fixed the bsinc SSE resamplers on non-GCC compilers.
37 Fixed support for extended capture formats with OpenSL.
39 Fixed handling of WASAPI not reporting a default device.
41 Fixed performance problems relating to semaphores on macOS.
43 Modified the bsinc12 resampler's transition band to better avoid aliasing
46 Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
48 Modified the virtual speaker layout for HRTF B-Format decoding.
50 Modified the PulseAudio backend to use a custom processing loop.
52 Renamed the makehrtf utility to makemhr.
54 Improved the efficiency of the bsinc resamplers when up-sampling.
56 Improved the quality of the bsinc resamplers slightly.
58 Improved the efficiency of the HRTF filters.
60 Improved the HRTF B-Format decoder coefficient generation.
62 Improved reverb feedback fading to be more consistent with pan fading.
64 Improved handling of sources that end prematurely, avoiding loud clicks.
66 Improved the performance of some reverb processing loops.
68 Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
69 some quality. Notably, down-sampling has less smooth pitch ramping.
71 Added support for SOFA input files with makemhr.
73 Added a build option to use pre-built native tools. For cross-compiling,
74 use with caution and ensure the native tools' binaries are kept up-to-date.
76 Added an adjust-latency config option for the PulseAudio backend.
78 Added basic support for multi-field HRTFs.
80 Added an option for mixing first- or second-order B-Format with HRTF
81 output. This can improve HRTF performance given a number of sources.
83 Added an RC file for proper DLL version information.
85 Disabled some old KDE workarounds by default. Specifically, PulseAudio
86 streams can now be moved (KDE may try to move them after opening).
90 Implemented capture support for the SoundIO backend.
92 Fixed source buffer queues potentially not playing properly when a queue
95 Fixed possible unexpected failures when generating auxiliary effect slots.
97 Fixed a crash with certain reverb or device settings.
101 Improved output limiter response, better ensuring the sample amplitude is
106 Implemented the ALC_SOFT_device_clock extension.
108 Implemented the Pitch Shifter, Frequency Shifter, and Autowah effects.
110 Fixed compiling on FreeBSD systems that use freebsd-lib 9.1.
112 Fixed compiling on NetBSD.
114 Fixed the reverb effect's density scale and panning parameters.
116 Fixed use of the WASAPI backend with certain games, which caused odd COM
117 initialization errors.
119 Increased the number of virtual channels for decoding Ambisonics to HRTF
122 Changed 32-bit x86 builds to use SSE2 math by default for performance.
123 Build-time options are available to use just SSE1 or x87 instead.
125 Replaced the 4-point Sinc resampler with a more efficient cubic resampler.
127 Renamed the MMDevAPI backend to WASAPI.
129 Added support for 24-bit, dual-ear HRTF data sets. The built-in data set
130 has been updated to 24-bit.
132 Added a 24- to 48-point band-limited Sinc resampler.
134 Added an SDL2 playback backend. Disabled by default to avoid a dependency
137 Improved the performance and quality of the Chorus and Flanger effects.
139 Improved the efficiency of the band-limited Sinc resampler.
141 Improved the Sinc resampler's transition band to avoid over-attenuating
144 Improved the performance of some filter operations.
146 Improved the efficiency of object ID lookups.
148 Improved the efficienty of internal voice/source synchronization.
150 Improved AL call error logging with contextualized messages.
152 Removed the reverb effect's modulation stage. Due to the lack of reference
153 for its intended behavior and strength.
157 Fixed resetting the FPU rounding mode after certain function calls on
160 Fixed use of SSE intrinsics when building with Clang on Windows.
162 Fixed a crash with the JACK backend when using JACK1.
164 Fixed use of pthread_setnane_np on NetBSD.
166 Fixed building on FreeBSD with an older freebsd-lib.
168 OSS now links with libossaudio if found at build time (for NetBSD).
172 Fixed an issue where resuming a source might not restart playing it.
174 Fixed PulseAudio playback when the configured stream length is much less
175 than the requested length.
177 Fixed MMDevAPI capture with sample rates not matching the backing device.
179 Fixed int32 output for the Wave Writer.
181 Fixed enumeration of OSS devices that are missing device files.
183 Added correct retrieval of the executable's path on FreeBSD.
185 Added a config option to specify the dithering depth.
187 Added a 5.1 decoder preset that excludes front-center output.
191 Implemented the AL_EXT_STEREO_ANGLES and AL_EXT_SOURCE_RADIUS extensions.
193 Implemented the AL_SOFT_gain_clamp_ex, AL_SOFT_source_resampler,
194 AL_SOFT_source_spatialize, and ALC_SOFT_output_limiter extensions.
196 Implemented 3D processing for some effects. Currently implemented for
197 Reverb, Compressor, Equalizer, and Ring Modulator.
199 Implemented 2-channel UHJ output encoding. This needs to be enabled with a
200 config option to be used.
202 Implemented dual-band processing for high-quality ambisonic decoding.
204 Implemented distance-compensation for surround sound output.
206 Implemented near-field emulation and compensation with ambisonic rendering.
207 Currently only applies when using the high-quality ambisonic decoder or
208 ambisonic output, with appropriate config options.
210 Implemented an output limiter to reduce the amount of distortion from
213 Implemented dithering for 8-bit and 16-bit output.
215 Implemented a config option to select a preferred HRTF.
217 Implemented a run-time check for NEON extensions using /proc/cpuinfo.
219 Implemented experimental capture support for the OpenSL backend.
221 Fixed building on compilers with NEON support but don't default to having
224 Fixed support for JACK on Windows.
226 Fixed starting a source while alcSuspendContext is in effect.
228 Fixed detection of headsets as headphones, with MMDevAPI.
230 Added support for AmbDec config files, for custom ambisonic decoder
231 configurations. Version 3 files only.
233 Added backend-specific options to alsoft-config.
235 Added first-, second-, and third-order ambisonic output formats. Currently
236 only works with backends that don't rely on channel labels, like JACK,
239 Added a build option to embed the default HRTFs into the lib.
241 Added AmbDec presets to enable high-quality ambisonic decoding.
243 Added an AmbDec preset for 3D7.1 speaker setups.
245 Added documentation regarding Ambisonics, 3D7.1, AmbDec config files, and
246 the provided ambdec presets.
248 Added the ability for MMDevAPI to open devices given a Device ID or GUID
251 Added an option to the example apps to open a specific device.
253 Increased the maximum auxiliary send limit to 16 (up from 4). Requires
254 requesting them with the ALC_MAX_AUXILIARY_SENDS context creation
257 Increased the default auxiliary effect slot count to 64 (up from 4).
259 Reduced the default period count to 3 (down from 4).
261 Slightly improved automatic naming for enumerated HRTFs.
263 Improved B-Format decoding with HRTF output.
265 Improved internal property handling for better batching behavior.
267 Improved performance of certain filter uses.
269 Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
270 extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
274 Implemented device enumeration for OSSv4.
276 Fixed building on OSX.
278 Fixed building on non-Windows systems without POSIX-2008.
280 Fixed Dedicated Dialog and Dedicated LFE effect output.
282 Added a build option to override the share install dir.
284 Added a build option to static-link libgcc for MinGW.
288 Fixed building with JACK and without PulseAudio.
290 Fixed building on FreeBSD.
292 Fixed the ALSA backend's allow-resampler option.
294 Fixed handling of inexact ALSA period counts.
296 Altered device naming scheme on Windows backends to better match other
299 Updated the CoreAudio backend to use the AudioComponent API. This clears up
300 deprecation warnings for OSX 10.11, although requires OSX 10.6 or newer.
304 Implemented a JACK playback backend.
306 Implemented the AL_EXT_BFORMAT and AL_EXT_MULAW_BFORMAT extensions.
308 Implemented the ALC_SOFT_HRTF extension.
310 Implemented C, SSE3, and SSE4.1 based 4- and 8-point Sinc resamplers.
312 Implemented a C and SSE based band-limited Sinc resampler. This does 12- to
313 24-point Sinc resampling, and performs anti-aliasing.
315 Implemented B-Format output support for the wave file writer. This creates
316 FuMa-style first-order Ambisonics wave files (AMB format).
318 Implemented a stereo-mode config option for treating stereo modes as either
319 speakers or headphones.
321 Implemented per-device configuration options.
323 Fixed handling of PulseAudio and MMDevAPI devices that have identical
326 Fixed a potential lockup when stopping playback of suspended PulseAudio devices.
328 Fixed logging of Unicode characters on Windows.
330 Fixed 5.1 surround sound channels. By default it will now use the side
331 channels for the surround output. A configuration using rear channels is
334 Fixed the QSA backend potentially altering the capture format.
336 Fixed detecting MMDevAPI's default device.
338 Fixed returning the default capture device name.
340 Fixed mixing property calculations when deferring context updates.
342 Altered the behavior of alcSuspendContext and alcProcessContext to better
343 match certain Windows drivers.
345 Altered the panning algorithm, utilizing Ambisonics for better side and
346 back positioning cues with surround sound output.
348 Improved support for certain older Windows apps.
350 Improved the alffplay example to support surround sound streams.
352 Improved support for building as a sub-project.
354 Added an HRTF playback example.
356 Added a tone generator output test.
358 Added a toolchain to help with cross-compiling to Android.
362 Implemented EFX Chorus, Flanger, Distortion, Equalizer, and Compressor
365 Implemented high-pass and band-pass EFX filters.
367 Implemented the high-pass filter for the EAXReverb effect.
369 Implemented SSE2 and SSE4.1 linear resamplers.
371 Implemented Neon-enhanced non-HRTF mixers.
373 Implemented a QSA backend, for QNX.
375 Implemented the ALC_SOFT_pause_device, AL_SOFT_deferred_updates,
376 AL_SOFT_block_alignment, AL_SOFT_MSADPCM, and AL_SOFT_source_length
379 Fixed resetting mmdevapi backend devices.
381 Fixed clamping when converting 32-bit float samples to integer.
383 Fixed modulation range in the Modulator effect.
385 Several fixes for the OpenSL playback backend.
387 Fixed device specifier names that have Unicode characters on Windows.
389 Added support for filenames and paths with Unicode (UTF-8) characters on
392 Added support for alsoft.conf config files found in XDG Base Directory
393 Specification locations (XDG_CONFIG_DIRS and XDG_CONFIG_HOME, or their
394 defaults) on non-Windows systems.
396 Added a GUI configuration utility (requires Qt 4.8).
398 Added support for environment variable expansion in config options (not
399 keys or section names).
401 Added an example that uses SDL2 and ffmpeg.
403 Modified examples to use SDL_sound.
405 Modified CMake config option names for better sorting.
407 HRTF data sets specified in the hrtf_tables config option may now be
408 relative or absolute filenames.
410 Made the default HRTF data set an external file, and added a data set for
411 48khz playback in addition to 44.1khz.
413 Added support for C11 atomic methods.
415 Improved support for some non-GNU build systems.
419 Fixed a regression with retrieving the source's AL_GAIN property.
423 Fixed device enumeration with the OSS backend.
425 Reorganized internal mixing logic, so unneeded steps can potentially be
426 skipped for better performance.
428 Removed the lookup table for calculating the mixing pans. The panning is
429 now calculated directly for better precision.
431 Improved the panning of stereo source channels when using stereo output.
433 Improved source filter quality on send paths.
435 Added a config option to allow PulseAudio to move streams between devices.
437 The PulseAudio backend will now attempt to spawn a server by default.
439 Added a workaround for a DirectSound bug relating to float32 output.
441 Added SSE-based mixers, for HRTF and non-HRTF mixing.
443 Added support for the new AL_SOFT_source_latency extension.
445 Improved ALSA capture by avoiding an extra buffer when using sizes
446 supported by the underlying device.
448 Improved the makehrtf utility to support new options and input formats.
450 Modified the CFLAGS declared in the pkg-config file so the "AL/" portion of
451 the header includes can optionally be omitted.
453 Added a couple example code programs to show how to apply reverb, and
456 The configuration sample is now installed into the share/openal/ directory
457 instead of /etc/openal.
459 The configuration sample now gets installed by default.