Sound Server Release Just Presented Pulse Audio 16.0, which acts as an intermediary between applications and various low-level sound subsystems, abstracting the work from the hardware.
PulseAudio peAllows you to control the volume and sound mix at the level of individual applications, organize the sound input, mix and output in the presence of several input and output channels or sound cards, allows you to change the format of the audio stream on the fly and the use of plugins, makes it possible to transparently redirect the audio stream to another machine.
Main new features of PulseAudio 16.0
In this new version it is highlighted that added the ability to use the Opus audio codec to compress the audio sent using the module-rtp-send module (previously only PCM was supported). To enable Opus, you must compile PulseAudio with GStreamer support and set the "enable_opus=true" setting in the module-rtp-send module.
Modules for transmitting/receiving sound through tunnels (tunnel-sink and tunnel-source) now have the ability to adjust the delay using the latency_msec parameter (previously, a delay of 250 microseconds was set).
It is also highlighted that there is a new algorithm to keep latency stable during resampling adaptive in the loopback of the module and elsewhere. Part of that is a new "time smoother" implementation. It will offer more accurate and stable latency estimates compared to the current algorithm. This is mainly important when a fixed relationship between different streams is required (A/V sync, modulo-loopback, modulo-combine-sink, modulo-echo-cancellation, …).
Since this is quite complex new code in the main audio processing parts, the old implementation is kept for a while to have a backup in case bugs appear.
Transmit/Receive Modules audio through tunnels provide support for reconnection to the server in case of connection failure. To enable reconnection, set the reconnect_interval_ms setting.
Parameter adjust_threshold_usec added to module-loopback module to adjust delay management algorithm (default delay is 250 microseconds). The default value of the adjust_time parameter has been reduced from 10 to 1 second, the possibility of setting values less than one second (for example, 0,5) has been added. Logging of playback speed settings is disabled by default and is now controlled by a separate log_interval option.
Added support for providing apps with battery level information of Bluetooth audio devices. The charge level is also displayed among the device properties shown in the "pactl list" output (property bluetooth.battery).
La ability to generate information in JSON format has been added to the pactl utility. The format is selected using the '–format' option, which can take the values text or json.
Added support for stereo output when using EPOS/Sennheiser GSP 670 headphones and SteelSeries GameDAC, which use separate ALSA devices for stereo and mono (previously only one mono device was supported).
Of the other changes that stand out in this new version:
- Fixed issues with reception from sound cards based on the Texas Instruments PCM2902 chip.
- Added Komplete 6-channel external sound card support Audio 6 MK2 from Native Instruments.
- Timing and accuracy issues in determining delays when transmitting audio through tunnels and the combined-sink module have been resolved.
- Added parameters sink_enabled and source_enabled to the module-jackdbus-detect module used to enable transmit/receive audio through JACK to selectively enable only transmit or receive audio through JACK. Module reloading is also allowed to use different JACK configurations at the same time.
- The remix parameter has been added to the module-combine-sink module to disable channel remixing, which may be necessary, for example, when using multiple sound cards to form a single surround sound.
Finally, if you are interested in being able to know more about it, you can consult the details in the following link