| Age | Commit message (Collapse) | Author | Files | Lines |
|---|
|
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.
|
|
All checksums have been double-checked against existing RMD160 and
SHA512 hashes.
The following distfiles couldn't be fetched (possibly they are fetched
conditionally):
./audio/freeswitch-sounds-ru/distinfo freeswitch/freeswitch-sounds-ru-RU-elena-32000-1.0.13.tar.gz
./audio/freeswitch-sounds-ru/distinfo freeswitch/freeswitch-sounds-ru-RU-elena-48000-1.0.13.tar.gz
./audio/freeswitch-music/distinfo freeswitch/freeswitch-sounds-music-32000-1.0.8.tar.gz
./audio/freeswitch-music/distinfo freeswitch/freeswitch-sounds-music-48000-1.0.8.tar.gz
./audio/freeswitch-sounds-fr/distinfo freeswitch/freeswitch-sounds-fr-ca-june-32000-1.0.18.tar.gz
./audio/freeswitch-sounds-fr/distinfo freeswitch/freeswitch-sounds-fr-ca-june-48000-1.0.18.tar.gz
./audio/freeswitch-sounds-en/distinfo freeswitch/freeswitch-sounds-en-us-callie-32000-1.0.22.tar.gz
./audio/freeswitch-sounds-en/distinfo freeswitch/freeswitch-sounds-en-us-callie-48000-1.0.22.tar.gz
|
|
|
|
Problem reported by wiz@.
|
|
|
|
|
|
* Convert to use meson to build.
Changelog:
1. Notes for end users
1. Support for LDAC and AptX bluetooth codecs, plus "SBC XQ" (SBC with
higher-quality parameters)
2. Support for HFP bluetooth profiles
3. Support for Bluetooth A2DP AVRCP Absolute Volume
4. ALSA path configuration files can now be placed in user home directory
5. module-virtual-surround-sink rewritten
6. More options for module-jackdbus-detect
7. Improved hardware support
1. SteelSeries Arctis 9
2. HP Thunderbolt Dock 120W G2
3. Behringer U-Phoria UMC22
4. OnePlus Type-C Bullets
5. Sennheiser GSX 1000/1200 PRO
8. New udev variable: PULSE_MODARGS
9. max_latency_msec argument added to module-null-source
10. module-filter-apply can take filter parameters from device properties
11. module-match can now be loaded multiple times
12. Improvements to FreeBSD support
13. Windows support added to Meson
14. Additional commands for pactl
15. Card profiles can be set to sticky
2. Notes for application developers
1. New API for sending messages from clients to PulseAudio objects
2. New mechanism for applications to disable shared memory on their
connection to PulseAudio
3. Notes for packagers
1. Autotools build system have been dropped
2. The startup script can now read additional configuration from the /etc/
pulse/default.pa.d/ directory
3. Option to build client library and utilities only
4. Avoid loading X11 modules on Wayland (GNOME-only for now)
5. OSS support is now configurable in Meson
6. Valgrind support is now configurable in Meson
|
|
|
|
|
|
|
|
|
|
PulseAudio 14.2
A bug fix release.
* Fix port switching when unplugging headphones
PulseAudio 14.1
A bug fix release.
* Support upto 8 mixer channels on ALSA devices
* Handle ALSA jacks with the same name but different index values
* Switch to plugged-in headset when mic availability is unknown
* Fix a potential segfault in the Bluetooth oFono HFP backend
* Fix a problem with module-ladspa-sink when avoid-resampling=true
* Fix database names containing canonical host for meson builds
|
|
Changelog:
git shortlog
Alexander E. Patrakov (1):
man: Deprecate the enable-remixing option
Alexander Patrakov (1):
Split the enable-lfe-remixing setting into two
Arun Raghavan (28):
rtp: Make init return a value for success/failure
rtp: Don't use cookie for SSRC
rtp: Drop support for non-L16 media
rtp: Move MTU handling to the RTP implementation
rtp: Hide RTP implementation details from module-rtp-*
rtpoll: Separate out before/after/work callback userdata
rtp: Add a GStreamer-based RTP implementation
rtp: Add some logging to know what backend is being used
switch-on-connect: Fix warning on discarded const qualifier
alsa-ucm: Support Playback/CaptureVolume
gitlab: Make container updates automatic
build: Bump soversion for libpulse
ci: Update for changes in CI template
ci: Go back to using CI templates from master
alsa-mixer: Add an explicit profile-set for Audigy devices
build-sys: Bump soversions
build-sys: Add doxygen/meson.build to distfiles
build-sys: Bump gettext dependency
Revert "alsa-mixer: support up to 8 channels per mixer element"
build-sys: meson: Add libm dep to raop module
rtp: gstreamer: Don't count on buffer DTS for capture time
rtp: gstreamer: Account for rounding errors in RTP timestamp conversion
build-sys: Set the GStreamer RTP backend to disabled by default
sink, source: Skip filter streams while changing default sink/source
switch-on-port-available: Switch to headphones on unknown availability
module-alsa-card: Drop availability groups with only one port
Revert "mainloop: fix timeout assignment in pa_mainloop_prepare"
build-sys: Bump libpulse soversion
Bal??zs Mesk?? (1):
Translated using Weblate (Hungarian)
Baurzhan Muftakhidinov (1):
i18n: Add initial Kazakh translation
Ben Buchwald (2):
module-jackdbus-detect: Allow omitting channels argument
module-jackdbus-detect: Separate sink/source channels arguments
Daniil Kovalev (1):
Fix memory leak in context_free
Dave Chiluk (1):
alsa-mixer: add support for LucidSound LS31, and create usb-gaming-headset profile
David Heidelberg (2):
meson: convert post-install.sh to python
meson: convert to install_headers
Dusan Kazik (1):
Translated using Weblate (Slovak)
Eero Nurkkala (3):
alsa-ucm: disallow null mdev argument into pa_alsa_open_mixer_by_name()
tests: hashmap-test.c: fix memory leak
tests: cpu-remap-test.c: fix memory leaks
Emanuil Novachev (1):
Translated using Weblate (Bulgarian)
Emilio Herrera (1):
Translated using Weblate (Spanish)
Felipe Sateler (2):
qpaeq: Drop unused imports
qpaeq: use python3 instead of python 2
Felix Yan (1):
shell-completion: zsh: Correct a typo
Geert Warrink (1):
Translated using Weblate (Dutch)
Georg Chini (5):
virtual sources: Propagate asyncmsgq change after source-output move
sink-input, source-output: Fix stream rescue if a move fails
daemon.conf: Add boolean rescue_streams parameter
sink, source: Fix stream rescue from sinks or sources without port
stream-restore: Restore preferred device for new streams
G?ran Uddeborg (1):
Translated using Weblate (Swedish)
Hugo Osvaldo Barrera (1):
Delete .travis.yml
Hui Wang (19):
sink-input: change bool save_sink to char *preferred_sink
sink-input: add a new API pa_sink_input_set_preferred_sink
sink-input: clear the preferred_sink if it is default_sink
core: move sink-inputs conditionally when update default_sink
sink: move streams to new appeared sinks if they prefer these sinks
device-port: moving streams due to changing the status of active_port
sink: move the streams to the default_sink when the sink is unlinked
stream-restore: skip entries setting action from gnome-control-center
source-output: change bool save_source to char *preferred_source
source-output: add a new API pa_source_output_set_preferred_source
source-output: clear the preferred_source if it is default_source
core: move source-outputs conditionally when update default_source
source: move streams to new appeared sources if they prefer these sources
device-port: moving streams since active_port on source changes status
source: move the streams to the default_source when the source unlink
stream-restore: skip entries set on source from gnome-control-center
alsa-mixer: store the ucm_device with the order of their priority
alsa: make the unsuspend more robust
alsa: adjust ucm sink/source priority according to ports priority
Igor V. Kovalenko (9):
module-stream-restore: log error writing volume/mute/device entry to database
module-stream-restore: check if dbus entry exists in dbus_entries map before creating it
device-port: fire port available changed hook after streams are moved
build-sys: meson: adjust path to gsettings-helper runing from build tree
module-bluez5-discover: avoid use after free on de-init
pactl: explicitly print if port availability is unknown
i18n: Update pulseaudio.pot
module-alsa-card: Add debug logging if availability group was pruned
switch-on-port-available: Add logging for port availability group
Jan Alexander Steffens (heftig) (5):
autotools: Put module-tunnel-source X11_CFLAGS into CFLAGS instead of LDFLAGS
meson: Define TUNNEL_SINK for module-tunnel-sink
build-sys: meson: Fix detection of SYS_memfd_create
meson: Add missing thread_dep to atomic-test
build: Make alsadatadir configurable
Jarno Suni (2):
shell-completions/bash: Add pactl set-default-sink and set-default-source
shell-completion: Do not use 'awk -e' in bash completion
Jaroslav Kysela (40):
alsa-ucm: use ucm2 name for the direct card index open
alsa-ucm: add mixer IDs to ucm_items
alsa-mixer: handle the index for ALSA mixer element identifiers
alsa-mixer: improve alsa_id_decode() function
alsa-ucm: use the correct mixer identifiers as first
alsa-ucm: add support for master volume
alsa-ucm: split correctly JackHWMute device names
alsa-ucm: fix parsing for JackControl
alsa-ucm: add comments to ucm_get_mixer_id()
alsa-ucm: validate access to PA_DEVICE_PORT_DATA()
alsa-ucm: parse correctly the device values
alsa-ucm: do not try to use UCM device name as jack name by default
alsa-util: do not try to guess the mixer name from the PCM name
alsa-ucm: add control and mixer device items
alsa-ucm: get the mixer names from ucm, don't guess
alsa-ucm: use the proper mixer name for ucm pcm sink/source
alsa-mixer: handle interface type (CARD,PCM) for mixer element lookups
alsa: rewrite mixer open/close, cache mixer accesses in probe
alsa-ucm: add support for HDMI ELD
alsa-mixer: do the quick card number lookup to save mixer instances
alsa-mixer: improve check for the empty path set for sink/source
alsa-ucm: allow to set profile priority from UCM value
alsa-ucm: correct the channel default logic (stereo)
alsa ucm: do not assign JackHWMute when JackControl is missing for the UCM device
ucm: fix the port / ucm device activation on boot
alsa sink/source: fix the mixer initialization
device-port: introduce available_group member
device-port: add type member
protocol: describe v34 (available_group, port type)
alsa-ucm: set available_group (use jack name)
alsa: legacy card - set available_group
alsa: legacy card - implement device port type parser and assignment
alsa ucm: set device port type
pactl: print device port type and available group
ucm: add possibility to skip the UCM card completely (Linked)
alsa: mixer - reorder the type field in path parser
alsa: fix analog-input-microphone-headset device type
alsa: fix type for legacy hdmi devices
alsa-mixer: support up to 8 channels per mixer element
alsa: move the exceptionally large value errors from error to debug level
Jaska Uimonen (1):
alsa-ucm: Fix volume control based on review
Jean-Baptiste Holcroft (1):
Translated using Weblate (French)
Josh (1):
alsa-mixer: add support for SteelSeries Arctis Pro 2019 headset
Juliano de Souza Camargo (1):
Update Portuguese translation
Kai-Heng Feng (3):
alsa: Skip resume PCM if hardware doesn't support it
alsa-mixer: Recognize USB audio jack mixer
module-alsa-card: Set a minimum profile priority if it's not set
Karl Ove Hufthammer (1):
Translated using Weblate (Norwegian Nynorsk)
Khem Raj (1):
remap/arm: Adjust inline asm constraints
Krzysztof Stasiowski (1):
alsa-mixer: Add support for SteelSeries Arctis 5 2019 headset
Laurent Bigonville (2):
alsa-mixer: Add the ability to pass the intended-role to the mapping
alsa-mixer: Set the intended-role of Steelseries Arctis 5/7 headset as phone
Libin Yang (2):
core-subscribe: add PA_SUBSCRIPTION_EVENT_CARD in dump_event
device-port: queue CARD CHANGE event before update default sink
Marc Ranolfi (1):
card-restore: prevent segfault caused by 'restore_bluetooth_profile=true'
Michael Pivonka (1):
alsa-mixer: Add Razer Kraken Tournament Edition USB headset
Milo Casagrande (2):
l10n: Update Italian translation
Translated using Weblate (Italian)
Milo Ivir (1):
Translated using Weblate (Croatian)
Nick Moriarty (1):
Permit root-owned home directory
O?uz Ersen (1):
Translated using Weblate (Turkish)
Pali Roh??r (5):
bluetooth: Implement reading SO_TIMESTAMP for A2DP source
bluetooth: Print SO_TIMESTAMP warning for SCO source only once
bluetooth: Ensure that only one A2DP codec is registered to bluez
bluetooth: policy: Remove BlueZ 4 related code
alsa: Fix compile warnings
Peter Levine (1):
atomic: Explicitly cast void* to unsigned long
Peter Meerwald (1):
macro: Move PA_LIKELY()/PA_UNLIKELY(), PA_CLAMP()/PA_CLAMP_UNLIKELY() to pulse/gccmacro.h
Philip Withnall (1):
daemon: Add --log-target=journal to pulseaudio.service
Piotr Dr?g (1):
Translated using Weblate (Polish)
RODRIGUEZ Christophe (1):
raop: Allow channel map module argument
Rafael Fontenelle (3):
i18n: Update Brazilian Portuguese translation
Update Brazilian Portuguese translation
Update Brazilian Portuguese translation
Ralph Seichter (1):
macos: Add missing import statement
Rasmus Thomsen (1):
meson: link libintl if it's not provided by libc
Rickie Schroeder (1):
start-pulseaudio-x11: fix KDE version check
Rosen Penev (2):
modules: fix wrong formats under 32-bit
raop-crypto: add missing header
Ryszard Knop (1):
switch-on-connect: Add blacklisting
Sanchayan Maity (2):
modules: rtp-gstreamer: Fix RTP sound lag
rtp: Fix sending of small packets
Sebastian Dr?ge (3):
rtp: Use yes/no in configure instead of 1/0
rtp: Use udpsink instead of fdsink for the GStreamer RTP implementation
rtp: Properly timestamp buffers in the GStreamer sender pipeline
Sebastien (1):
Update configure.ac to fix the enable_gstreamer summary
StefanBruens (1):
raop: Send initial timing packet to prime UDP connection tracking
Taahir Ahmed (1):
Add a basic test suite for pa_hashmap
Tanu Kaskinen (56):
Update NEWS
alsa-mixer-path-test: Hide unused functions when building with Meson
daemon-conf: disable flat-volumes by default
null-sink: allow clients to configure the supported formats
alsa-mixer: recognize the "Speaker Jack" control
man: remove outdated information about real-time scheduling
gitlab: explain the container image tag better
stream: clarify the meaning of NULL device
core-util: Handle zero-length volume string
switch-on-connect: Blacklist HDMI devices by default
virtual-source: Don't process the rewind during requesting it
virtual-surround-sink: Use correct sample spec with memblockq
sink, source: Fix inaccurate log message
alsa: Document that mixer elements can be identified by a combination of name and index
ucm: Don't log errors during normal operation
protocol-native: Fix error code
i18n: Import Bulgarian translation from fedora.zanata.org
i18n: Merge Catalan translation from fedora.zanata.org
i18n: Replace po/ja.po with the Fedora Zanata version
i18n: Replace po/es.po with the Fedora Zanata version
i18n: Add pulseaudio.pot to version control
meson: Remove libpulse dep from libpulsecore
Point to SupportedAudioFormats wiki page where appropriate
thread-mainloop: Extend the locking documentation a bit
stream-restore: Drop the version field from the entry struct
stream-restore: Fix a potential crash in pa_namereg_is_valid_name()
stream-restore: Forget pre-14.0 stream routing
raop-sink: Fix compiler warnings
sink, source: Use the global configuration for the avoid_resampling default
man: Explain the limitations of "pulseaudio --check"
alsa-ucm: Fix NULL pointer handling
i18n: Update pulseaudio.pot
build-sys: meson: bump protocol version
remap_neon: use register r12 instead of r7
device-port: send sink and source change events when availability changes
build-sys: Build atomic-test only if pthread_setaffinity_np() is available
meson: Build atomic-test
udev: fix too long card name with HyperX Cloud Orbit S
gitignore: Ignore hashmap-test
build-sys: Configure doxygen.conf.in a bit differently
meson: Add doxygen target
Improve the port available_group and type documentation
alsa-mixer: Fix mapping_group_available() logic
Rename "available group" to "availability group"
alsa-mixer: Fix indentation
i18n: Update pulseaudio.pot
man: client.conf: Explain that autospawn=no doesn't disable systemd autostarting
alsa-mixer: Document the intended-roles mapping option
ci: Fix comment
alsa-mixer: Set availability groups once per card
alsa-mixer: Fix jack name comparison
gitlab: Remove trailing whitespace from the issue template
alsa-mixer: Remove references to non-existent multichannel paths
alsa-mixer: Document the description and description-key mapping options
switch-on-port-available: Fix switching away from unplugged headphones
build-sys: Disable GStreamer by default with Autotools
Timo Gurr (1):
meson: allow to disable installing completions
Tom Yan (2):
main, core: check idle after loading conf
man: mention that exit-idle-time is complied if the user is lingering
Tomasz Kontusz (4):
alsa-mixer: Fix well-known descriptions for steelseries game/chat outputs
alsa-mixer: Remove unused iec958-passthrough* descriptions
alsa-mixer: add description-key to Mappings and Profiles
alsa-mixer: add support for Astro A50 gaming headset
Vasilis Tsiligiannis (1):
start-pulseaudio-x11: Make 'plasma' version check shell portable
Wim Taymans (5):
X11: Add xauthority parameter
alsa: handle unavailbale HW volume in UCM
alsa-ucm: use the right profile name
modules: fix some small memory leaks
alsa-util: fix check for digit
Yi-Jyun Pan (1):
l10n: zh_TW: update translation
Yuri Chornoivan (1):
i18n: Update poulseaudio.pot
ckdo (8):
raop: Fix non working tcp mode
rtp: Fix reverted test for INHIBIT_AUTO_SUSPEND_ONLY_WITH_NON_MONITOR_SOURCES
raop: Fix soft volume not applied on initial volume
raop: Fix rewinding handling : process just after request received
raop: Fix callback call in raop client after auth : only call once everything is freed
raop: Code clarification : Detect raop_client recording state in a proper way
raop: Code clarification : Only free the I/O thread when everything is initialized
raop: Add autoreconnect feature
efim (1):
add comma
itsthem (1):
pulsecore: Replace gendered pronouns with gender neutral ones
muzena (1):
i18n: Update the Croatian translation
roshal (1):
man: remove space
zhaochengyi (1):
pulsecore: Add exception judgment to fix the crash when playing music
|
|
Bump PKGREVISION for NetBSD binary change.
|
|
Changelog:
PulseAudio 13.0
Changes at a glance:
* Added support for Dolby TrueHD and DTS-HD Master Audio
* Improved initial card profile selection for ALSA cards
* Bluetooth card profile choices aren't persistent any more by default
* Added support for SteelSeries Arctis 5 USB headset
* New "max_latency_msec" module argument for module-loopback
* New "stream_name" module argument for module-rtp-send
* Fixed S/PDIF for CMEDIA USB2.0 High-Speed True HD Audio
* Use source sample spec and channel map by default in module-loopback
* New "avoid_resampling" module argument for module-udev-detect and module-alsa-card
* "avoid_resampling" also tries to avoid format conversion if the ALSA device supports it
* New function to enable realtime scheduling for client threads
* Removed BlueZ 4 support
* Dropped intltool
* Introduction of the Meson build system
* Const-ification of parameters across headers
* Minor bug-fixes, bindings updates and several translation updates
|
|
|
|
Bump PKGREVISION.
|
|
|
|
PulseAudio 12.2
The previous release tarball contained a broken configure script, this release
fixes the tarball. There are no changes in the source, except for this NEWS
file update.
PulseAudio 12.1
A bug fix release.
* Fixed crash when switching to A2DP bluetooth profile
* Fixed plugin search path in module-ladspa-sink
* Fixed file permissions for the pipes created by module-pipe-sink and
module-pipe-source
|
|
Changelog:
PulseAudio 12.0 release notes
Changes at a glance
* Better latency reporting (and hence better A/V sync) with the A2DP
bluetooth profile
* Much more accurate latency reporting for AirPlay devices
* Fixed a crash or high CPU use problem with Intel HDMI LPE
* module-switch-on-connect now ignores virtual devices
* When using passthrough for compressed audio, set the "non-audio" bit
* Prioritize HDMI output over S/PDIF output
* HSP support for more bluetooth headsets
* Choose the A2DP bluetooth profile by default instead of HSP
* New "sink_input_properties" module argument for module-ladspa-sink
* New "use_system_clock_for_timing" module argument for module-pipe-sink
* module-pipe-sink can now use an existing pipe
* Steelseries Arctis 7 USB headset stereo output support
* Dell Thunderbolt Dock TB16 speaker jack support
* Fixed digital input support for some USB sound cards
* Fixed Native Instruments Traktor Audio 6 detection
* Ability to disable input or output on macOS
* New "dereverb" option for the Speex echo canceller
* New module: module-always-source
* State files not any more readable by all users in the system mode
* module-augment-properties now uses XDG_DATA_DIRS to find .desktop files
* Updates for the Vala bindings
* The GConf dependency can now be avoided
* qpaeq license changed from AGPL to LGPL
* qpaeq ported to Qt 5
* Compatibility with glibc 2.27
* The esdcompat tool isn't any more installed if esound support is disabled
|
|
From Santhosh Raju in PR pkg/53381
|
|
avoid defining a duplicate symbol already found in glibc.
from upstream commit:
https://github.com/pulseaudio/pulseaudio/commit/dfb0460fb4743aec047cdf755a660a9ac2d0f3fb
From Nia Alarie in PR pkg/53305
|
|
Changelog:
* Fix a crash in filter modules related to flat volumes and volume
sharing
* Fix a crash when the bluetooth adapter reports weird MTU size
* Disable bluetooth MTU autodetection by default
* Add mixer handling back for hardware that doesn't have any alsa-lib
configuration
* Prioritize USB devices over built-in sound cards (11.0 was supposed
to have this feature, but the implementation turned out to be
incomplete)
|
|
|
|
Changelog:
PulseAudio 11.0 release notes
Changes at a glance
Support for newer AirPlay hardware
USB and bluetooth devices preferred over internal sound cards
The default sink and source configuration is remembered better
Bluetooth HSP headset role implemented
Bluetooth HFP audio gateway role implemented (requires oFono)
Bluetooth HSP audio gateway and HFP hands-free unit roles can be enabled simultaneously
Upmixing can now be disabled without bad side effects
Avoid having unavailable sinks or sources as the default
Option to avoid resampling more often
Option to automatically switch bluetooth profile to HSP more often
Better latency regulation in module-loopback
Changed module argument names in module-ladspa-sink and module-virtual-surround-sink
Fixed input device handling on Windows
Improved bluetooth MTU configuration (warning! this causes some hardware to not work any more, see the details below for how to fix it)
GNU Hurd support
Applications can request LADSPA or virtual surround filtering for their streams
Support for 32-bit applications on 64-bit systems in padsp
|
|
Changelog:
# PulseAudio 10.0 release notes
## Changes at a glance
* Automatically switch Bluetooth profile when using VoIP
applications
* New module for prioritizing passthrough streams
(module-allow-passthrough)
* Fixed hotplugging support for USB surround sound cards
* Separate volumes for Bluetooth A2DP and HSP profiles
* memfd-based shared memory mechanism enabled by default
* Removed module-xenpv-sink
* Dropped dependency to json-c
* When using systemd to start PulseAudio, pulseaudio.socket is
always started first
* Compatibility with OpenSSL 1.1.0
* Clarified qpaeq license
## Notes for end users
### Automatically switch Bluetooth profile when using VoIP applications
Bluetooth headsets typically support both the A2DP profile, which is
suitable for music, and the HSP profile, which is suitable for
telephony use cases. module-bluetooth-policy will now automatically
switch the profile of a Bluetooth headset from A2DP to HSP/HFP when an
application creates a recording stream with property media.role=phone
(telephony applications should set that property for their
streams). When the stream goes away, the profile gets restored back to
A2DP. This way the user doesn't have to manually switch the profiles
when starting and stopping a call. This behaviour can be disabled by
giving argument auto_switch=false to module-bluetooth-policy.
### New module for prioritizing passthrough streams (module-allow-passthrough)
Passthrough streams are streams whose content must go completely unaltered from the application to the sound card, and they are mainly used for playing back compressed audio through an S/PDIF connection. When a passthrough stream is playing to a device, no other streams can play at the same time to the same device, and if there's already something playing when a passthrough stream is created, the passthrough stream creation will normally fail. Now we have a new module, called module-allow-passthrough, which will give higher priority to passthrough streams, so that if there are other streams playing when a passthrough stream is created, those other streams will be moved out of the way to a dummy device.
### Fixed hotplugging support for USB surround sound cards
A long-standing bug that prevented PulseAudio from using any
hotplugged USB surround sound cards has been fixed.
### Separate volumes for Bluetooth A2DP and HSP profiles
The Bluetooth sink and source names are now different depending on
whether the active profile is A2DP or HSP. This allows PulseAudio to
store different volumes for A2DP and HSP. Due to different mechanisms
for implementing volume control between the profiles, trying to use
the same volume value in PulseAudio doesn't actually result in the
same perceived volume, so it's better not to try to use the same
volume.
As an unfortunate transition effect, volumes saved earlier with an
older PulseAudio version won't have effect in the new PulseAudio
version, so all Bluetooth devices will have their volume reset to the
default value when running the new PulseAudio version for the first
time.
This only affects BlueZ 5 users. The same change was done already
earlier for PulseAudio's BlueZ 4 code.
### memfd-based shared memory mechanism enabled by default
The memfd-based shared memory mechanism that was implemented in 9.0 is
now enabled by default. This shouldn't cause any user-visible changes
in behaviour, except if you're using the Firejail sandboxing software,
which doesn't work with PulseAudio's old shared memory mechanism.
If desired, the feature can still be disabled by setting "enable-memfd
= no" either in daemon.conf (for disabling it at the server side) or
in client.conf (for disabling it at the client side).
### Removed module-xenpv-sink
module-xenpv-sink was removed, because it's probably not used by
anyone. If you use it, please let us know.
## Notes for packagers
### Dropped dependency to json-c
libpulse previously used json-c internally, which forced applications
to link to json-c too. That caused crashing in some GLib applications,
because json-c and json-glib both use the same name for some
functions. To solve this, we implemented the necessary JSON
functionality directly in libpulse, so we don't depend on json-c any
more.
### When using systemd to start PulseAudio, pulseaudio.socket is always started first
The pulseaudio.service unit now depends on pulseaudio.socket, meaning
that before systemd starts PulseAudio, it will always first set up the
socket. This is done to avoid confusing behaviour in certain corner
cases (see the comments in pulseaudio.service for a more detailed
explanation).
### Compatibility with OpenSSL 1.1.0
OpenSSL, which is used by module-raop-sink, broke backwards
compatibility in the 1.1.0 release. PulseAudio now supports both 1.1.0
and older versions.
### Clarified qpaeq license
Most of PulseAudio is licensed under LGPL, but the "qpaeq" equalizer
GUI is licensed under AGPL. That hasn't previously been mentioned
anywhere else than in the qpaeq source code itself. Therefore,
distributions that tag their packages with license information have
likely used incorrect information. The license of qpaeq hasn't
changed, but the use of AGPL is now correctly noted in the top-level
LICENSE file.
|
|
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
|
|
* Based on wip/pulseaudio by kamil@
Changelog:
Automatic routing more likely to change profile
OS X and NetBSD support improvements
Systemd journal logging for clients
New LFE balance programming interface
Module-dbus-protocol improvements
More flexible configuration file handling
pulsecore-8.0.so moved to a private directory
New script for measuring memory consumption
Various bug fixes and small improvements
|
|
|
|
Avoid to multiple declaration by typedef.
|
|
|
|
Changelog:
Since we had a couple of annoying bugs in 7.0, we thought it'd be a
good idea to do a 7.1 to address those.
Changes at a glance:
* Fix a crasher when using srbchannel
* Fix a build system typo that caused symlinks to turn up in /
* Make Xonar cards work better
* Other minor bug fixes and improvements
|
|
Changelog:
PulseAudio 7.0
Changes at a glance:
* LFE channel synthesis with low-pass filtering
* New libsoxr based resamplers
* Socket activation support for TCP
* The "srbchannel" IPC mechanism enabled by default
* More flexible jack detection support when using UCM
* Exiting due to SIGTERM isn't considered a failure
* Better support for Creative SoundBlaster Omni Surround 5.1
|
|
|
|
modifiers. It is beyond common sense why pulseaudio devs considered the
mechanical conversion to inline asm an improvement...
|
|
Previous fix by tnn@ requires devel/gettext-tools.
It is too heavy.
|
|
* Remove obsolete hal option for PLIST.
Changelog:
PulseAudio 6.0 Release Notes
Changes at a Glance
BlueZ 5 native HSP (headset) support
BlueZ 5 HFP (hands-free) profile support via oFono
systemd socket activation support
Better support for multichannel and 2.1 profiles
Remap optimisations
Many minor improvements, bug fixes, and i18n updates
Notes for Application Developers
New function in libpulse: pa_stream_write_ext_free(). The function allows more flexible use of free callbacks than the regular pa_stream_write() function. This is useful if an audio buffer is part of a bigger structure that needs to be freed or unreferenced when the audio buffer is no longer needed.
We now have Vala bindings for libpulse-simple.
|
|
|
|
|
|
mkpatches refresh of the previously existing patches
|
|
|
|
Use /dev/audio instead of /dev/sound under NetBSD.
This change fixes unstable audio output.
O.k. by wiz@.
|
|
|
|
on Linux. What were they thinking... If there is one platform that it's
rash to make assumptions about API consistency on, it's Linux...
Fixes build on Ubuntu 11.10.
|
|
* Remove dependency to sysutils/hal, hald backend is removed.
* Use deprecated OSS backend under NetBSD (alsa backend should be used?)
Changelog:
PulseAudio 5.0
Changes at a glance:
* BlueZ 5 support (A2DP only)
* Reimplementation of the tunnel modules
* Native log target support for systemd-journal
* Resampler refactoring
* --monitor-stream option for parecord and parec
* "latency_msec" argument for module-rtp-recv
* "inhibit_auto_suspend" argument for module-rtp-send
* "auto" argument for module-tunnel-sink and module-tunnel-source
* Removed module-bluetooth-proximity
* Jack detection for line out
* Laptop internal surround speaker volume support
* Improved float->s16 and s16->float sample conversion for ARM NEON
* "Available" flag for card profiles
* Removed module-dbus-protocol from the default configuration
* Lots of other enhancements, bug fixes, and documenation and i18n updates
Detailed change log:
http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/5.0
PulseAudio 4.0
Changes at a glance:
* Better handling of low latency requests
* Optimisations while mixing (generic, ARM NEON)
* Default resampler is now speex-float-1 (lower CPU usage)
* Major Bluetooth refactoring for better reliability and easier maintenance
* Fixes for graceful hand-off to/from JACK
* New module to apply ducking based on stream roles
* Echo canceller infrastructure fixes
* Bash and zsh completion for command line tools
* Solaris and OS X fixes
* Lots of other enhancements, bug fixes, and documenation and i18n updates
Detailed change log:
http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/4.0
PulseAudio 3.0 Release Notes
We're, back with another shiny PulseAudio release! While the 3.0 release was a little delayed, it brings a number of important improvements, and bug fixes. A summary of changes follows.
Notable Changes
ALSA Use Case Manager (UCM) support
Runtime editable LADSPA filter parameters
Out-of-the-box support for Bluetooth sources
ARM NEON optimisations
Configurable device latency offset
Adhere to the XDG Base Directory Specification
Various ALSA changes
Lots of infrastructure improvements
Packaging
Bluetooth support requires now "sbc", a library for the SBC codec. The codec used to be included within PulseAudio, but it has now been split off into a separate library. It's available at http://www.bluez.org.
Support for the "socket API" of BlueZ has been dropped in favour of the D-Bus based "media API". Due to this change, the minimum supported version of BlueZ is now 4.99. Also, make sure that you don't have "Disable=Media" in /etc/bluetooth/audio.conf. And due to a bug in BlueZ, it's probably necessary to have "Disable=Socket", otherwise there will be problems with the A2DP profile.
Support for HAL has been removed. This shouldn't affect anyone, but if it does, please configure PulseAudio to use udev instead. module-hal-detect still exists for maintaining configuration file compatibility, but all it does is to load module-udev-detect. module-hal-detect may get completely removed in 4.0.
|
|
|
|
|
|
Based on PR 48405 by Nat Sloss.
|
|
|
|
|
