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of the Discrete Fourier Transforms library.
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Changelog:
PulseAudio 9.0
Changes at a glance:
* Automatic routing improvements
* Beamforming and various other new features in the WebRTC echo canceller
* Various improvements in module-role-cork and module-role-ducking
* LFE remixing disabled by default
* memfd-backed shared memory transport
* Support for sample rates up to 384 kHz
* webrtc-audio-processing dependency minimum version bumped to 0.2
* Changed the C standard from C99 to C11.
Detailed change log:
http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/9.0
Contributors
Ahmed S. Darwish
Alexander E. Patrakov
Arun Raghavan
Barun Kumar Singh
David Henningsson
Deepak Srivastava
Gabor Kelemen
Georg Chini
Jeremy Huddleston Sequoia
Jonathan Perkin
Juho Hämäläinen
Jungsup Lee
Kamil Rytarowski
Marcin Lewandowski
Milo Casagrande
Muhammet Kara
Nazar Mokrynskyi
Peter Meerwald
Piotr Drąg
Sachin Kumar Chauhan
Sangchul Lee
Tanu Kaskinen
YunQiang Su
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Major new features:
- conditionals in format strings, with syntax %{?CONDITION?A[?B]}
- sndio output plugin (OpenBSD)
- VTX (libayemu) input plugin
- optional mouse support (:set mouse=true to enable)
Notable changes:
- lots of new formatting options:
- format_treewin
- format_treewin_artist
- format_trackwin_album
- format_statusline
- play count support (%X or %{play_count} in format strings)
- Bash completion support for cmus and cmus-remote
- support for CMUS_SOCKET environment variable
- format_print remote command
- pwd command
- auto_expand_albums option is split into
- auto_expand_albums_follow
- auto_expand_albums_search
- auto_expand_albums_selcur
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libopenmpt 0.2-beta19 (2016-07-23)
[Change] libopenmpt now uses C++14 [[deprecated]] attribute
instead of compiler-specific solutions when appropriate.
[Change] libopenmpt C++ header now uses C++11 noexcept instead
of C++98 throw() exception specification when supported. throw()
is deprecated since C++11. This does not change API or ABI as
they are equivalent. Use LIBOPENMPT_ASSUME_CPLUSPLUS_NOEXCEPT
to override the default.
[Change] The preprocessor macro LIBOPENMPT_ANCIENT_COMPILER_STDINT
is gone. Please use LIBOPENMPT_ASSUME_CPLUSPLUS_CSTDINT instead.
Additionally, the typedefs moved from illegal namespace ::std
into somewhat less dangerous namespace ::openmpt::std. You can
test #ifdef LIBOPENMPT_QUIRK_NO_CSTDINT client-side to check
whether libopenmpt.hpp used the non-standard types. (Note: Of
all supported compilers, this change only affects the 3 compilers
with only limited support: MSVC 2008, GCC 4.1, GCC 4.2.)
[Bug] xmp-openmpt: Crash when viewing sample texts.
The public libopenmpt C++ header has auto-detection logic for
the used C++ standard now. In case your client code compiler
misreports the standard version or you want to override it for
other reasons, #define LIBOPENMPT_ASSUME_CPLUSPLUS to the value
of the standard version you desire to be used. There is also a
macro for each individual aspect, like
LIBOPENMPT_ASSUME_CPLUSPLUS_CSTDINT,
LIBOPENMPT_ASSUME_CPLUSPLUS_DEPRECATED,
LIBOPENMPT_ASSUME_CPLUSPLUS_NOEXCEPT which take precedence over
the general macro.
Portamento with sample swap behaviour was wrong for ProTracker
MODs.
Rewritten loader and various playback fixes for MDL files.
libopenmpt 0.2-beta18 broke import of many pattern commands in
DBM, DMF and ULT files.
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Add pulseaudio option, default on to follow upstream.
libopenmpt 0.2-beta18 (2016-07-11)
[Change] openmpt123: Add PulseAudio output support. Autotools and
Makefile build systems now depend on libpulse and libpulse-simple
by default. Disable with --without-pulseaudio or NO_PULSEAUDIO=1
respectively. When enabled, PulseAudio will be the default output
driver,
[Change] xmp-openmpt: Settings are now stored in xmplay.ini like
with every other plugin.
[Regression] openmpt123: Support for FLAC < 1.3.0 has been removed.
FLAC before 1.3.0 is broken beyond repair as it provides assert.h
in the include path.
[Bug] Generated pkg-config file libopenmpt.pc by both Makefile and
Autotools build systems was totally broken.
[Bug] libopenmpt no longer uses the non-thread-safe global std::rand()
function.
[Bug] Sample loops in GDM modules did not work when using Emscripten.
[Bug] XM and MO3 loaders could crash due to unaligned memory accesses.
[Bug] Fixed incorrect handling of custom MPTM tunings on big endian
platforms.
[Bug] Fixed various problems found with clang 3.8 static analyzer,
address sanitizer and undefined behaviour sanitizer.
[Bug] File header probing functionality was broken for most formats.
[Bug] With non-seekable streams, the entire file was almost always
cached even if it was not of any supported module type.
Seeking in allsubsongs-mode now works correctly.
openmpt123: Added subsong support.
Various playback fixes for 669, IT, MT2 and MTM files.
Some MOD files with more than 128 patterns (e.g. NIETNU.MOD) were
not loaded correctly.
A new example libopenmpt_example_c_probe has been added which
demonstrates the usage and flexibility of openmpt_could_open_propability()
in the C API under various constraints.
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This Opus 1.1.3 release focuses mainly on optimizations and bug
fixes. Changes include:
* Neon optimizations improving performance on ARMv7 and ARMv8 by up to 15%
* Fixes some issues with 16-bit platforms (e.g. TI C55x)
* Fixes to comfort noise generation (CNG)
* Documenting that PLC packets can also be 2 bytes
* Includes experimental ambisonics work (--enable-ambisonics)
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Version 1.3.2 -- July 9, 2016
=============================
- Fixed crash on an invalid audio file that FFmpeg could not decode.
- Fixed build on Ubuntu 14.04 with libav.
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Speech Dispatcher:
Key features:
* Common interface to different TTS engines
* Handling concurrent synthesis requests -- requests may come
assynchronously from multiple sources within an application and/or
from different applications
* Subsequent serialization, resolution of conflicts and priorities of
incomming requests
* Context switching -- state is maintained for each client connection
independently, event for connections from within one application
* High-level client interfaces for popular programming languages
* Common sound output handling -- audio playback is handled by Speech
Dispatcher rather than the TTS engine, since most engines have limited
sound output capabilities
What is a very high level GUI library to graphics, Speech Dispatcher is
to speech synthesis. The application neither needs to talk to the devices
directly nor to handle concurrent access, sound output and other tricky
aspects of the speech subsystem.
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--------------------------------
ver 0.19.17 (2016/07/09)
* decoder
- flac: fix assertion failure while seeking
- flac: fix stream duration indicator
- fix seek problems in several plugins
* fix spurious seek error "Failed to allocate silence buffer"
* replay gain: fix "replay_gain_handler mixer" setting
* DSD: use 0x69 as silence pattern
* fix use-after-free bug on "close" and "kill"
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-----------------------------
Version 3.9:
* support full covariance for 'gmmp' command.
* add -B, -c1 and -c2 option to 'gmmp' command to specify block size and structure of covariance matrix.
* change temporary file path for csh.
* bug fixes.
Version 3.8:
* add 'c2ndps' command for transform from cepstrum to negative derivative of phrase spectrum.
* add 'mgclsp2sp' command for transform from LSP to spectrum.
* add 'ndps2c' command for transform from negative derivative of phrase spectrum to cepstrum.
* add 'wavjoin' command to save stereo wav file.
* add 'wavsplit' command to load stereo wav file.
* add -V option to 'dtw' command for loading DTW results.
* add -e option to calculate inverse matrix stably for 'delta' and 'vc' commands.
* support MAP estimation in 'gmm' command.
* replace -i option to -q option for 'lsp2lpc,' 'lspcheck,' and 'mgclsp2mgc' commands.
* change -c option of 'mlsacheck' command.
* update swipe and hts_engine API.
* bug fixes.
Version 3.7:
* add 'vc' command to perform GMM-based voice conversion.
* add 'lsp2sp' command to transform LSP coefficients to spectrum.
* add -R option to 'mlsacheck' command to specify threshold value.
* add -f option to 'gmmp' command to specify full covariance matrix.
* add -M, -c1 and -c2 option to 'gmm' command to specify block size and structure of covariance matrix.
* add -n option to 'delta' command to specify order of regression polynomial.
* add -magic and -MAGIC option to 'rmse' command to delete and substitute magic number.
* add -n and -t option to 'rmse' command to specify order and number of vector, respectively.
* add -POWX, -LOGX option, and new constant (semitone) to 'sopr' command.
* add -l option to 'decimate' and 'interpolate' command to specify length of vector.
* add -g and -G option to 'lspcheck' command to check value of gain.
* add -L option to 'lspcheck' command to specify log gain.
* add -n option to 'mgc2mgclsp' command to specify split number of unit circle.
* add -p and -d to 'mgc2mgclsp' command to specify maximum number and end condition of interpolation.
* add -x 16, 22, 24 and 48 option of 'glogsp' and 'grlogsp' command to specify more x-axis scales.
* add filter coefficients for 7:4 down-sampling, 3:5 and 7:8 up-sampling.
* unify option to specify log gain into -L option.
* fix -c option of 'mlsacheck' command to specify the type of modification of filter coefficients.
* bug fixes.
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The latter allows users to override the value with a newer version.
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Version 1.0.27 (2016-06-19)
* Fix an SF_INFO seekable flag regression introduced in 1.0.26.
* Fix potential infinite loops on malformed input files.
* Add string metadata read/write for CAF and RF64.
* Add handling of CUE chunks.
* Fix enaian-ness issues in PAF files.
* Minor bug fixes and improvements.
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Ardour is a multi-channel digital audio workstation, allowing users to record,
edit, mix and master audio and MIDI projects. It is targeted at audio
engineers, musicians, soundtrack editors and composers.
Originally packaged in pkgsrc-wip by:
- myself,
- <coypu>,
- Thomas Klausner.
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liblo is an implementation of the Open Sound Control
protocol for POSIX systems.
Originally packaged in pkgsrc-wip by:
- Johann Franz,
- Sergey Svishchev,
- Blair Sadewitz,
- Tobias Nygren,
- Joerg Sonnenberg,
- Thomas Klausner,
- Aleksej Saushev,
- Blue Rats
and
- myself.
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and OpenGL as sane options for output for now. Bump revision.
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ver 0.19.16 (2016/06/13)
* faster seeking
* fix system include path order
* add missing DocBook file to tarball
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xjadeo is a simple video player that gets sync from jack.
Originally packaged in pkgsrc-wip by myself.
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No binary change is intended.
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* fosshub.com does not provide direct link for distfile
Changelog:
2.1.2
Improvements
Effects
Noise Reduction defaults changed.
Interface
We've made several improvements to the spectrogram view of tracks which are particularly valuable for vocal work.
New option 'Spectral Reassignment'. This algorithm deduces a 'finer' spectrogram for vocal work.
Four new Spectrogram scales.
Improvement to the Pitch (EAC) algorithm - cleaner (sharper) display.
Spectrogram Settings are now available per-track.
Other Changes
Upgraded from wxWidgets 2.8.12 to wxWidgets 3.0.2.
Restructuring of the Preferences Menu
Restructuring of the Track Dropdown Menu (for the Spectrograms)
The Noise Reduction "Frequency smoothing" default has been increased to 3 to help avoid metallic artifacts. Resetting Preferences or deleting audacity.cfg will set the new default.
(Windows) The location of Audacity's work-in-progress folder has been changed, to avoid problems where certain disk cleaners treated them as deletable temporary files.
Bug Fixes
Crashes
Crashes on reading .wav files with many channels (CVE-2016-2540) and corrupt .mp2 files (CVE-2016-2541) have been fixed. Thanks to Chris Navarrete from Fortinet's FortiGuard Labs for notifying us of these bugs and associated security risk.
2.1.1
Improvements
Effects
Noise Reduction defaults changed.
Interface
We've made several improvements to the spectrogram view of tracks which are particularly valuable for vocal work.
New option 'Spectral Reassignment'. This algorithm deduces a 'finer' spectrogram for vocal work.
Four new Spectrogram scales.
Improvement to the Pitch (EAC) algorithm - cleaner (sharper) display.
Spectrogram Settings are now available per-track.
Other Changes
Upgraded from wxWidgets 2.8.12 to wxWidgets 3.0.2.
Restructuring of the Preferences Menu
Restructuring of the Track Dropdown Menu (for the Spectrograms)
The Noise Reduction "Frequency smoothing" default has been increased to 3 to help avoid metallic artifacts. Resetting Preferences or deleting audacity.cfg will set the new default.
(Windows) The location of Audacity's work-in-progress folder has been changed, to avoid problems where certain disk cleaners treated them as deletable temporary files.
Bug Fixes
Crashes
Crashes on reading .wav files with many channels (CVE-2016-2540) and corrupt .mp2 files (CVE-2016-2541) have been fixed. Thanks to Chris Navarrete from Fortinet's FortiGuard Labs for notifying us of these bugs and associated security risk.
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iSuil is a lightweight C library for loading and wrapping LV2 plugin
UIs.
Suil makes it possible to load a UI of any toolkit in a host using
any other toolkit (assuming the toolkits are both supported by
Suil). Hosts do not need to build against or link to foreign toolkit
libraries to use UIs written with that toolkit; all the necessary
magic is performed by dynamically loaded modules. The API is designed
such that hosts do not need to explicitly support specific toolkits
at all - if Suil supports a particular toolkit, then UIs in that
toolkit will work in all hosts that use Suil automatically.
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The SoX Resampler library 'libsoxr' performs one-dimensional
sample-rate conversion - it may be used, for example, to resample
PCM-encoded audio.
It aims to give fast and high quality results for any constant
(rational or irrational) resampling ratio. Phase-response, preserved
bandwidth, aliasing, and rejection level parameters are all
configurable; alternatively, simple `preset' configurations may be
selected. An experimental, variable-rate resampling mode of operation
is also included.
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From Piotr Meyer.
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