![]() ![]() I've done lots of digging, and they've been my endgame choices for awhile now. I highly recommend DeaDBeeF or Audacious as graphical music players.While it's not bit perfect, there are times when up-sampling the bit depth to 32 bit while maintaining a faithful sample rate between track changes would be close enough in accordance to my convenience-related needs. Though, it's worth mentioning that making PulseAudio avoid re-sampling is the only made that has no foreseeable CPU/RAM performance drawbacks (it actually improves CPU performance). I recommend this leddit post and this medium post to achieve audibly better audio fidelity from PulseAudio. ![]() Especially is the case when dealing with multiple streams of different sample rates, when you don't feel like messing with a new distro install too much, etc. ![]() While PulseAudio will likely never technically be able to be configured to be bit perfect, it will likely be quite a deal more convenient than most other alternatives for awhile longer.Exactly, what I would like to being able to avoid for certain use cases. *Unless of course some resampling is bridged in between. Currently I am not even shure, wether the "-rates" setting is also applicable to recording. And that I wish to be able to configure with pipewire as well.ītw. So there certainly seem to exist some mechanisms to forcing an application into a specific sample rate. ![]() And so far, the detection of the samplerate required by jack has been working fine. To name jackd, that by design is running with a fixed samplerate, and the applcation cannot change this*. However, what I have been looking for is a way to tell pipewire not to resample, so if the application request a samplerate that is not available, because the stream is coming in with a different samplerate, to report an (unsupported) error to the application. And for most use cases that transparency is perfectly fine. If both do not match, pipewire will resample to satisfy the application. I'd rather suspect, the application makes a request for a given samplerate, and pipewire then brokers between what actually comes in (if it is a direct digital input where the AD converter sets the clock) or tries to set the samplerate accordingly (if it is an anlog input or some digital/digital converters). I am not entirely sure, that is the full story. What comes as the input is defined by the application starting the recording Thanks again, I will see, what I can do and get back with more information. Should have checked that earlier, but was fixed on the pw-top infos. Anyway, I guess, I need to sort this out first, even though I am not yet sure where to start.Īnalog to USB on the DrDAC works and as fas as I understand the manual, SPDIF (coax & optical) ->USB should work as well. Audacity confirms no level, but I never got along with that program, so that may just be me. However, for my principle problem: Actually looking at the pw-record file, it is empty, so somehow so signal arrives at all. Just opening it is enough, no need to change any settings. And after stopping pw-record pw-top only switches back to 44.1kHz as soon as I open the mixer (plasma-pa, does again not happen with pulsemixer, though). Pw-top shows an input of 44.1 kHz, however, when I then run pw-record without any arguments, it switches to the default 48kHz. As there are practically no USB->toslink Soundcards available (as in toslink -> USB, other way round is no problem). I am trying to feed an analog signal, converted by a RME ADI-2 ADC via toslink into an ESI DrDAC, which in turn should convert the toslink signal into USB. Indeed it seems, I have a different problem. ![]()
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