Age | Commit message (Collapse) | Author | Files | Lines |
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Can be removed on next update.
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The revisions were not reset on version upgrade and the committer
name was wrong in CHANGES. Please excuse the retro-fixup.
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Highlighted bugfixes:
Security fixes in Matroska, MP4 and AVI demuxers
Fix scrambled video playback with hardware-accelerated VA-API decoders on certain Intel hardware
playbin3/decodebin3 regression fix for unhandled streams
Fragmented MP4 playback fixes
Android H.265 encoder mapping
Playback of MXF files produced by older FFmpeg versions
Fix rtmp2sink crashes on 32-bit platforms
WebRTC improvements
D3D11 video decoder and screen recorder fixes
Performance improvements
Support for building against OpenCV 4.6 and other build fixes
Miscellaneous bug fixes, memory leak fixes, and other stability and reliability improvements
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Fix pkglint.
* ALSA and WinMIDI drivers now pass system real-time messages on to user callback (#1115, thanks to @albedozero)
* Fix FPU division by zero in fluid_player_set_tempo() (#1111)
* Fix system-wide config file not loaded (#1118)
* Pluseaudio driver now honors audio.periods setting (#1127, thanks to @pedrolcl)
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1.2.7.2
Core
Release v1.2.7.2
Mixer API
mixer: add documentation about postcondition of removal event processing
PCM API
pcm: share plugin: handle -EINTR
pcm: share plugin: return error if socket read or write call fails
RawMidi API
rawmidi: fix the params_mode check condition in snd_rawmidi_tread()
Use Case Manager API
ucm: fix st_mode check for symbolic links
Kernel Headers
include/sound/type_compat.h: fix include guard
1.2.7.1
Core
Release v1.2.7.1
conf: Use ino64_t to save and compare inode numbers
Control API
control: eld - fix the decoding for older hw
I/O API
output: include stdarg.h
PCM API
pcm: dmix - Add error handler for `fgets`
Use Case Manager API
ucm: list also hardware configs (hw:X) in uc_mgr_scan_master_configs() fcn
Configuration
conf: Use ino64_t to save and compare inode numbers
1.2.7
Core
Release v1.2.7
configure: remove --with-lfs option, but keep the autodetection code
configure: add --with-lfs option
gitcompile: fix 32 bit compilation support
remove .travis.yml (using github actions)
github actions: move to checkout@v3
conf: fix the export of safe_strto* functions from libasound
Config API
ucm: add ${evali:} substitution
Control API
control: eld - add missing ctype.h header inclusion
control: shm - initialize write buffer
control: decode HDMI device name from ELD
PCM API
pcm: hw: change rate range syntax
pcm: hw: add "min_rate" and "max_rate" as alternatives to single "rate" parameter
pcm: rate - rewrite the may_wait_for_avail_min callback for the rate plugin
pcm: plugin - fix avail_min calculation on rate plugin
pcm: dmix: fix wrong scaling in 32bits pcm mixing
pcm: ladspa - Use LFS calls (readdir64)
pcm: fix for the unitialized write buffer
control: decode HDMI device name from ELD
pcm: multi: return correct hwptr and avail from snd_pcm_multi_status()
pcm: direct - allow 'off' string for hw_ptr_alignment
pcm: direct - cleanups for snd_pcm_direct_reset_slave_ptr()
pcm: direct - add support for channel bindings in snd_pcm_direct_query_chmaps()
pcm: direct: Check xrun/suspend before the slave hwptr update
pcm: direct: Move slave PCM state checks into XRUN check helper
pcm: direct: Improved suspend/resume support
pcm: direct: Propagate error code from snd_pcm_direct_client_chk_xrun()
pcm: rate: fix drain of partial period at end of buffer
Topology API
src/topology/parser.c: drop duplicate safe_strtol_base
Use Case Manager API
ucm: fix the reload call (snd_use_case_mgr_reload)
ucm: implement disdevall sequence command
use-case.h: add Channels/ChannelPos values to the documentation
ucm: fix memory leak in the error path (Include)
ucm: Use LFS calls (stat, scandir)
ucm: main - fix the compilation error (signess)
ucm: macro - make argument names shorter
ucm: doc - describe variants, minor corrections
ucm: implement enadev2 and disdev2 sequence commands
ucm: add support for verb variants
ucm: set SYNTAX_VERSION_MAX to 6
ucm: move macros and evali substitution to Syntax 6
ucm: macro - add deep call protection (recursion)
ucm: macro call inplace evaluate inside macro
ucm: macro - fix the error message, print id
ucm: return empty string for undefined "open" variables (arguments)
ucm: allow '-' prefix to avoid errors when the variable is not defined
ucm: allow passing variables through ucm open string
use-case.h: add SND_USE_CASE_DEV_DIRECT define
ucm: implement MacroDefine and Macro subtree evaluation
ucm: local_config may be NULL (error path)
ucm: add ${evali:} substitution
ucm: fix the '${eval:EXPR}' substitution
ucm: top-level path - set directory from symlink
ALSA Server
pcm: fix for the unitialized write buffer
Configuration
conf: Use LFS calls when reading config files
conf: vc4-hdmi: use a proper hdmi pcm, fix broken default pcm
conf: fix memory leak in snd_config_substitute() for strings
conf: snd_config_merge - fix comment (overwrite / override)
conf: fix the export of safe_strto* functions from libasound
Documentation
README: Add link to GitHub Actions
Simple Abstraction Mixer Modules
mixer: simple module: python 3.10 PyTuple_SET_ITEM() fix
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Upstream changes since last pkgsrc version:
1.30.1
------
- mpg123:
-- Show stderr of network helpers in -vvv mode.
-- Use curl --http0.9, if available, to support shoutcast v1 streams
without wget (wget not needing such switch, yet).
-- Support file:// URLs for local access as was intended with the last
release.
-- Give more helpful error message if neither wget nor curl are usable, also
allow error messages from curl to appear when not --quiet.
-- Update the man page.
1.30.0
------
- build:
-- Use dummy as default module when no other outputs are enabled. This also
fixes a non-module build with just the dummy (bug 333).
-- Use CMAKE_CURRENT_SOURCE_DIR in CMake build to help nested use (bug 335).
-- some updates for OS/2 support (fixing up stdin playing, for example)
- mpg123:
-- new network backend using external tools/libraries to also support HTTPS
-- old network backend changed to use h_addr_list[0] instead of h_addr
-- terminal control keys now case-sensitive (fixing smal/big pitch controls)
-- additional terminal control keys for simple equalizer control (A/a for bass,
J/j for mids, N/n for treble, e for reset, E for printout)
-- terminal volume control now in decibel steps and bounded to +/- 60 dB
-- terminal control now also with audio from stdin (bug 338) via
/dev/tty or ctermid()
-- terminal control also available for OS/2 and Windows platforms
-- re-print tag info on decrease of terminal width for a bit less mess
-- always print an empty line after tag info for cleaner appearance
-- print lyrics also to stderr
-- remote control API v10 with "@P 3" as additonal message on track end
-- also added PROGRESS command as opposite of SILENCE
-- fix some verbosity, tweak help for --icy-interval
-- added --auth-file
-- also obscure argument to --auth for others
-- Cygwin/MinGW: Provide _win32_utf8_wide and _win32_wide_utf8 unconditionally.
It is needed by the WASAPI plugins, the underlying conversion functions
should be present since Windows 2000. Fixes WASAPI support on Cygwin.
Also needed for new network code.
- libout123:
-- pulse: initialize more error codes to avoid bogus error messages
-- os2: considerable fixup for proper writes of full buffers avoiding
nasty effects from the ... special audio system, more cleanup still
nice-to-have, but still lacking
1.29.4
------
- libmpg123:
-- Saturate reader file position at off_t limit to satisfy
undefined behaviour checkers.
-- Avoid harmless unitialized value in ID3v1 check (filepos, later being
set before actual use).
- libout123:
-- Build fix for win32_wasapi output for predefined _WIN32_WINNT (bug 329),
thanks to Vincent Torri.
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pkgsrc changes:
---------------
* Add directory to find the reside-builder library in sidplay2.
* Bump revision.
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ver 0.23.8 (2022/07/09)
* storage
- curl: fix crash if web server does not understand WebDAV
* input
- cdio_paranoia: fix crash if no drive was found
- cdio_paranoia: faster cancellation
- cdio_paranoia: don't scan for replay gain tags
- pipewire: fix playback of very short tracks
- pipewire: drop all buffers before manual song change
- pipewire: fix stuttering after manual song change
- snapcast: fix busy loop while paused
- snapcast: fix stuttering after resuming playback
* mixer
- better error messages
- alsa: fix setting volume before playback starts
- pipewire: fix crash bug
- pipewire: fix volume change events with PipeWire 0.3.53
- pipewire: don't force initial volume=100%
* support libfmt 9
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5.5.0:
fixed: CVE-2021-44269 (encoding crafted DSD file triggers OOB read crash)
fixed: very long filenames cause stack-overflow crash in all CLI programs
fixed: the length stored in WAV headers not always corrected when using -i
fixed: attempting to encode raw DSD audio from stdin sometimes causes crash
fixed: DSD to PCM decimation: small clicks between tracks and tiny DC offset
fixed: length update in library-generated WAV headers on big-endian machines
fixed: sanitize custom extensions read from WavPack files to be alphanumeric
added: accepting brace-delimited options in the wavpack executable filename
added: "--drop" option to Windows executables for multi-file "drag-and-drop"
added" "--raw-pcm" option to wvunpack executable (does DSD --> 24-bit PCM)
added: "--no-overwrite" option to wavpack executable (to resume sessions)
improved: build system clean-up including switch to non-recursive "make"
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What's Changed
Add session configs in #60
Add browse page in #61
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1.2.2
Fix #65: new crash when missing `meta` parameter
Older changes not found.
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[v0.6.18] Released on: July 8th, 2022.
New: Add lqueue and tqueue function similar to cmus.
New: include theme files in binary because I saw they are not included in the
aur package.
New: Fetch invidious instance from website, so that they'll not expire and
search youtube will always works.
Fix: When playing mp3 encoded by iTunes under gapless mode, symphonia backend
will panic.
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v0.9.6 (2020-12-28) : True Blue
New
Id3.Tag(version=) keyword argument.
Expose TextFrame ctor kwargs to Apple frames.
Added --about CLI argument for extra version/program info.
Fix
Preserve linked file info in Tag.clear().
Handle v1 .id3/.tag files.
Improved art plugin behavior when missing dependencies.
[art plugin] Improved error for missing dependencies.
TYER conversion (and restored non v2.2 breakage, for now)
ID3 v2.2, date getters return values again.
Passed filtered files list or handleDirectory, and skip non-existant symlinks
Fixed installation supported Python text.
Implement v1.0/v1.1 tag conversion rules.
Other
Poetry build system
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The changes from 1.22.1 include:
Fixed PipeWire version check.
Fixed building with PipeWire versions before 0.3.33.
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[v0.6.17]
Released on: July 6th, 2022.
-New: Search for database. Triggered by / key when focusing database.
-New: Gapless playback for symphonia/mpv/gstreamer backend. Toggle by Ctrl+g
and enabled by default.
-Fix: Youtube download mirrors are all broken. Replace them with new mirrors.
-Fix: After download from youtube, the prompt message will not disappear if
error happens.
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config values for pkgsrc-based installations
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script
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Snapcast is a multiroom client-server audio player, where all clients are
time synchronized with the server to play perfectly synced audio. It's not
a standalone player, but an extension that turns your existing audio player
into a Sonos-like multiroom solution.
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openal-soft-1.22.1:
Fixed CoreAudio capture.
Fixed air absorption strength.
Fixed handling 5.1 devices on Windows that use Rear channels instead of
Side channels.
Fixed some compilation issues on MinGW.
Fixed ALSA not being used on some systems without PipeWire and PulseAudio.
Fixed OpenSL capturing noise.
Fixed Oboe capture failing with some buffer sizes.
Added checks for the runtime PipeWire version. The same or newer version
than is used for building will be needed at runtime for the backend to
work.
Separated 3D7.1 into its own speaker configuration.
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Changes not found.
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Changes not found
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Changelog:
16.1:
The 16.0 release had some regressions, so here comes a bugfix
release to remedy those (along with some other fixes). These are
the changes:
* Fix parsing of percentage volumes with decimal points in pactl
* Fix crash with the "pacmd play-file" command when reads from
the disk aren't frame-aligned
* Fix module-rtp-recv sometimes thinking it's receiving an Opus
stream when it's not
* Fix frequent crashing in module-combine-sink, regression in 16.0
* Fix crashing on 32-bit architectures when using the GStreamer
codecs for LDAC and AptX
16.0:
## Notes for end users
Opus support in the RTP modules
The audio sent with module-rtp-send can now be compressed with the
Opus codec. To use it, pass enable_opus=true as a module argument
to module-rtp-send. This feature works only when PulseAudio is
compiled with GStreamer enabled (both sending and receiving end).
Stereo output support for EPOS/Sennheiser GSP 670 USB/wireless
headset and SteelSeries GameDAC
The EPOS/Sennheiser GSP 670 headset has separate mono and stereo
output ALSA devices, but with the default configuration only mono
worked with PulseAudio. Now both outputs work. The support includes
both direct USB connection and the GSA 70 wireless dongle.
The same fix was applied to SteelSeries GameDAC.
Fix input issues for Texas Instruments PCM2902 based sound cards
Texas Instruments PCM2902 is a generic audio chip that is used in
multiple USB sound cards. We had custom configuration for Behringer
UMC22, which turned out to affect multiple sound cards because they
use the same USB ID. The PCM2902 sound cards vary in their
capabilities, while our configuration was tailored only for the
UMC22 card, which caused some trouble with recording on multiple
PCM2902 sound cards. The reported issues have now been fixed.
Native Instruments Komplete Audio 6 MK2 profiles
The Native Instruments Komplete Audio 6 MK2 is similar to the
Komplete Audio 6 and is now supported as well.
Tunnel latency is now configurable
The tunnel sink and source modules used to have a fixed 250 ms
latency. The desired latency can now be configured with the
latency_msec module argument.
Tunnel modules can now reconnect to remote server
A new reconnect_interval_ms argument was added to all four tunnel
sink and source modules. When the argument is specified, the tunnel
module will try automatic re-connection to the remote server if
the connection fails. The argument specifies the time interval in
ms after which a connection attempt is repeated. In particular,
this allows to load tunnel sinks and sources from default.pa which
will become available as soon as the remote server becomes available.
Bluetooth device battery level reporting added
If a bluetooth device supports battery level reporting, PulseAudio
now is able to forward the information to other software. In case
your desktop environment doesn't yet support showing the battery
level in a nice GUI, the level is also available in the device's
card object properties with the bluetooth.battery key. The property
can be read with pactl list cards, for example.
Tunnel and combine-sink latency fixes
The tunnel and combine-sink latency reporting accuracy has been
improved, which should help with audio synchronization issues.
module-loopback improvements
As part of a set of improvements to module-loopback's latency
stability, a new argument, adjust_threshold_usec, was added to
module-loopback to fine-tune the controller algorithm. The default
value is 250 (microseconds), which should be sufficient in most
cases. If it's not enough (caused by inaccurate latency reports
from the sink or source), the loopback's sample rate will oscillate,
while unnecessarily high values will increase variance in the
loopback latency.
Another change is the ability to set the adjust_time argument to
smaller values than 1 second, for example 0.5 sets the adjustment
interval to half a second. The default value was changed from 10
seconds to 1 second to make the latency control tighter.
module-loopback used to log a bunch of status information every
time it adjusted the playback rate. Now that the default adjustment
interval is down from 10 seconds to 1 second, the logging became
a bit too much, and the logging was disabled by default. It can
now be enabled by setting the log_interval module argument. The
value is given in seconds, it doesn't have to be an integer. The
logging still happens at the time the rate adjustment is done, so
if log_interval is less than adjust_time, then the logging will
happen once per adjustment cycle.
Increased flexibility for module-jackdbus-detect
module-jackdbus-detect is used for loading a JACK sink and source
when JACK starts up. The module now has new sink_enabled and
source_enabled arguments that accept boolean values. The new
arguments can be used to disable either the sink or the source if
loading both is not desired.
module-jackdbus-detect can now also be loaded more than once,
allowing multiple JACK sinks or sources with different configurations
to be created.
pactl can show information in JSON format
pactl has a new option --format, which accepts values text and
json. text shows the pactl output in the traditional way, json
shows it in the JSON format for easier interfacing with other
software. Channel remixing can be disabled for module-combine-sink
module-combine-sink now accepts a boolean remix argument, which
can be used to disable normal remixing. This is useful when combining
multiple sound cards for surround output: if there are 3 stereo
sound cards, you might want to set the channel map of one card to
front-left,front-right, another to rear-left,rear-right and the
third to front-center,lfe. If a combine sink is then created with
a 5.1 surround channel map using these sound cards as slaves, audio
is copied to all these sound cards, but by default the audio is
downmixed to stereo for each card, which doesn't result in proper
s is done, the channels that don't fit the slave channel map are
just dropped, which means that each sound card gets audio only for
the intended channels.
## Notes for application developers
Stream latency reports now include resampler delay
Sink input and s, respectively. While this is minor semantic change,
it should allow for more accurate A/V sync for applications.
Bluetooth device battery level reporting added
If a bluetooth device supports battery level reporting, the level
is now reported to BlueZ. Aroperties with the bluetooth.battery
key. There are no notifications when the property value changes,
however (bug reported: #1314).
## Notes for packagers
Module installation location changed, remember to upgrade paprefs
to the latest version!
Modules are now installed to $libdir/pulseaudio/modules, previously
they were installed to $libdir/pulse-$version/modules. paprefs has
some logic that is sensitive to the module installation path, so
if you ship paprefs in your distribution, make sure to upgrade
paprefs to version 1.2. Earlier paprefs versions won't work properly
with PulseAudio 16.0.
Opus support in the RTP modules requires enabling GStreamer
The new Opus compression is available only when PulseAudio is built
with the gstreamer Meson option enabled (previously it was disabled
by default, now it's automatically enabled if the necessary
dependencies are found).
Bluetooth battery level reporting via BlueZ requires enabling
experimentals features in BlueZ
The Battery API is still marked as an experimental feature in BlueZ,
and if you wish to have PulseAudio use it, bluetoothd has to be
started with the --experimental command line argument.
New time smoother implementation
There's a new algorithm for keeping latency stable during adaptive
resampling in module-loopback and elsewhere. Part of that is a new
"time smoother" implementation. It will deliver more accurate and
stable latency estimations compared to the current algorithm. This
is mainly important where a fixed relationship between different
streams is required (A/V sync, module-loopback, module-combine-sink,
module-echo cancel, ...). Since this is a fair bit of complex new
code in the core audio processing parts, the old implementation is
kept around for a while to have a backup in case bugs show up. The
new time smoother can be disabled with the enable-smoother-2=false
Meson option.
Possibility to build the daemon without the client parts
It's now possible to build the daemon without building the client
parts at the same time, by using the -Dclient=false Meson option.
The daemon will still need the client libraries during the build,
the libraries installed in the system will be used. Apparently this
kind of scheme is useful for Gentoo.
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Shairport Sync is an AirPlay audio player - it plays audio streamed from
Apple devices and AirPlay sources such as ForkedDaapd (but apparently not
rtunes).
Audio played by a Shairport Sync-powered device stays synchronised with the
source and hence with similar devices playing the same source. In this way,
synchronised multi-room audio is possible for players that support it.
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Bump PKGREVISION again.
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What's Changed
-Cleanup compilation warnings in #56
-Remove user:{user} part when parsing URIs in #58
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ffmpeg5
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Remove distinfo for patch that isn't there.
Bump PKGREVISION.
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Bump PKGREVISION.
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doesn't provide any options.
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Changes since v1.54
v1.55 - 14.06.2022
- Fixed crash when using "Copy smp." on an empty dest. or source instrument
- Fixed: Using "Copy Ins." on an empty source instrument resulted in a
non-sensical system message.
- Reset pattern loop states on "Play Song", fixes a potential bug
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