What resampler in CDSP do you use?Squeezelite/Sox seems to be far more efficient at resampling than CDSP is
What resampler in CDSP do you use?Squeezelite/Sox seems to be far more efficient at resampling than CDSP is
Squeezelite/Sox seems to be far more efficient at resampling than CDSP is. IIRC the system load averages below 0.5 and maximum utilisation of each core around 25%. CDSP load reported in the GUI is less than 2%. CDSP has no problem dealing with the 192KHz here, but I'm only doing crossover filtering and gain. By contrast, when resampling 44.1KHz CD audio in CDSP with the same filters I have to limit it to 96KHz to avoid buffer underruns.
Yes I resample with this recipe: -R "vX:::28:95:105:45"
AsyncSinc Balanced at 96KHz on a Pi4b.What resampler in CDSP do you use?
The input clipping I was referring to is on the UR23 interface from the CD player, which doesn't use Squeezelite. The UR23 is straight 16/44.1 PCM into CDSP.It is the overall attenuation I meant rather than the lowpass filter. Looking at the squeezelite manpage, attenuation is the third field (after the second colon), and leaving it empty would apply the default attenuation of 1 dB. Maybe that is not enough for some "hot" recordings, thus the clipping you are seeing. You could test some offending audio files on the command-line. (I'm not sure what these technical considerations actually mean in terms of audible distortion.)
Async resampler is significantly more demanding than the sync one, used e.g. in SoX. Do you need the async version to assist the rate adjust?AsyncSinc Balanced at 96KHz on a Pi4b.
Yes, good point! I use async because the CD player/UR23 has a different clock to the M4 DAC. Otherwise the clocks drift and create buffer underruns/overruns.Async resampler is significantly more demanding than the sync one, used e.g. in SoX. Do you need the async version to assist the rate adjust?
No problem if you are lucky and the cd player runs faster than the DAC, then you just get a slowly increasing delay. If it's the other way around I wouldn't expect it to be possible.playing around with the buffer and chunk size to see if CDSP can play a whole CD without encountering buffering issues?
IIUC that hat uses only gpio pins for I2S and I2C (which is shared by principle) https://github.com/Ysurac/raspberry...t/dts/overlays/hifiberry-digi-pro-overlay.dtsAs all the gpio is occupied by the digi pro
Set the allowed sample rates on the capture device in pipewire, fixed sample rate to the loopback sink and then link them with pw-link.Next step is figuring out how to best do resampling in pipewire.
Any pointers?
alsa-sink:monitor_FL
alsa-sink:monitor_FR
alsa_output.platform-snd_aloop.0.analog-stereo:monitor_FL
alsa_output.platform-snd_aloop.0.analog-stereo:monitor_FR
alsa_input.platform-snd_aloop.0.analog-stereo:capture_FL
alsa_input.platform-snd_aloop.0.analog-stereo:capture_FR
alsa-sink:playback_FL
alsa-sink:playback_FR
alsa_output.platform-snd_aloop.0.analog-stereo:playback_FL
alsa_output.platform-snd_aloop.0.analog-stereo:playback_FR
pw-link -l
alsa-sink:playback_FL
|<- alsa_input.platform-snd_aloop.0.analog-stereo:capture_FL
alsa-sink:playback_FR
|<- alsa_input.platform-snd_aloop.0.analog-stereo:capture_FR
alsa_input.platform-snd_aloop.0.analog-stereo:capture_FL
|-> alsa-sink:playback_FL
alsa_input.platform-snd_aloop.0.analog-stereo:capture_FR
|-> alsa-sink:playback_FR