The release of the PulseAudio 16.0 sound server has been announced, which acts as an intermediary between applications and various low-level sound subsystems, abstracting hardware operations. PulseAudio allows for volume and sound mixing at the level of individual applications, organizes the input, mixing, and output of sound with multiple input and output channels or sound cards, enables on-the-fly changes to the sound stream format and the use of plugins, and allows for transparent redirection of the sound stream to another machine. The PulseAudio code is distributed under the LGPL 2.1+ license. It supports operation on Linux, Solaris, FreeBSD, OpenBSD, DragonFlyBSD, NetBSD, macOS, and Windows.
Key improvements in PulseAudio 16.0:
- The ability to use the Opus audio codec for compressing sound sent via the module module-rtp-send has been added (previously, only PCM was supported). To enable Opus, it is necessary to build PulseAudio with GStreamer support and set the configuration 'enable_opus=true' in the module module-rtp-send.
- In the modules for transmitting/receiving sound through tunnels (tunnel-sink and tunnel-source), the ability to configure delay using the parameter latency_msec has been added (previously, a delay of 250 microseconds was hardcoded).
- In the modules for transmitting/receiving sound through tunnels, support for automatic reconnection has been ensured in case of a connection drop. proxy server To enable reconnection, the reconnect_interval_ms configuration option should be set.
- Support has been added for providing applications with information about the battery level of Bluetooth sound devices. The battery level is also displayed among the device properties shown in the output of 'pactl list' (property bluetooth.battery).
- The pactl utility has been enhanced to output information in JSON format. The format selection is made using the ‘—format’ option, which can be set to either text or json.
- Support for stereo output when using EPOS/Sennheiser GSP 670 headsets and SteelSeries GameDAC has been added, for which separate ALSA devices are used for stereo and mono (previously, only a mono device was supported).
- Issues with receiving sound from sound cards based on the Texas Instruments PCM2902 chip have been resolved.
- Support for the 6-channel external sound card Native Instruments Komplete Audio 6 MK2 has been added.
- Issues with synchronization and accuracy of delay detection during audio transmission through tunnels and the combine-sink module have been resolved.
- The module-loopback has been enhanced with the adjust_threshold_usec parameter to fine-tune the delay management algorithm (the default delay is set at 250 microseconds). The default value for the adjust_time parameter has been reduced from 10 seconds to 1 second, allowing for values less than a second (for instance, 0.5). Logging of playback speed adjustment is now disabled by default and can be controlled via the separate log_interval option.
- The module-jackdbus-detect, used to activate audio transmission/reception via JACK, has been updated with sink_enabled and source_enabled parameters that allow selective enabling of audio transmission or reception through JACK. The module can also now be reloaded to facilitate simultaneous use of different JACK configurations.
- A parameter remix has been added to the module-combine-sink to disable channel remixing, which may be necessary when using multiple sound cards to create a unified surround sound experience.
Source: opennet.ru
