Hello
I record my band in live usb streaming with a qu-24 allen & heath ==> ok
I export the tracks.
But the key and beat are lower than the real recording (1/2 ton) after export.
I have to turn up the tone and the beat after every export.
I use jack / Alsa and Ardour 6.3.0 Kubuntu Linux
Where do you think this problem comes from?
Best regards
Looks like you were recording at 48kHz sampling rate and playing back at 44.1.
thank you for your reply anahata
I had checked this. All my sessions was recorded at 44.1 khz. BUT 
by reopening my session this time without the USB and by setting the frequency to 48 khz (with ARDOUR warning), I found the right format.
this is therefore a good palliative. thank’s
Can you explain to me then:
I record in USB at 44.1 khz ==> ok
I read in USB at 44.1 khz ==> ok
I export in USB ==> bad beat and tone reading
I read outside of USB (with my internal soundcard) ==> bad beat and tone
unless I open my session Ardour outside of USB in 48 khz
or else quite simply, whatever the frequency that one chooses, the USB recording will be in all cases at 48 khz?
A specificity of the QU-24?
Most likely the Qu24 is running at 48k (Fairly standard with a lot of digital consoles in that price range, on some you can change this on the console by adjusting the internal clock), which on Linux will mean your session actually runs at 48k even if you may think it is running at 44.1
SeabladeOK Seablade.
thank you both for these explanations.
And happy New Year
Problem solved
While Ardour is running, run the following command in a Terminal window
cd /tmp && wget https://community.ardour.org/files/adevices.sh && bash ./adevices.sh
It will not modify your system; it lists all soundcards, their current hardware settings and applications using them, etc. Then post the output here.
Chances are that even though the Ardour assumes the session is at 44.1kHz, the soundcard is running at 48kHz.
As long as you only use Ardour this is not easily noticeable since the issue is symmetric. record -> play uses the same rate. Only the playhead and clocks move a bit faster compared to wall-clock, and if you were to use MIDI synths they’d be out of tune…
nevermind, a quick web-search found the following:
The Qu is running at fixed 48kHz
from https://community.allen-heath.com/forums/topic/sample-rate
and
The Qu mixers have a sample rate of 48 Khz. and can not be changed.
Make sure that whatever DAW software you are using has the same sample rate of 48 kHz
from https://community.allen-heath.com/forums/topic/usb-b-streaming-audio-distortion-qu-24#post-53613
In your case with the A&H QU-24, the answer is yes.
Ardour doesn’t complain when the requested session rate and the hardware rate do not match?
Only if Ardour can know about this.
The issue is usually caused by JACK. jackd silently falls back to the nearest available rate without informing jack clients about it.
In this case it might also be that the driver reports 44.1kHz, even though the device runs at 48kHz (which is why I’ve asked for the adevices.sh output).
Hello everybody
Indeed the QU-24 works only in 48K and jack makes Ardour believe that we are in 44K (thank you Robin).
For my table, I just have to tell Ardour 48K when the session is started and everything is ok (export or restart the session recorded without USB).
Other: by modifying the frequency with QJACKCTL and restart, it does not change anything.
Again, thank you very much
cd /tmp && wget https://community.ardour.org/files/adevices.sh && bash ./adevices.sh
USB with ardour 44K
Card 1 (QU24):
Playback Device 0 (USB Audio):
Recording Device 0 (USB Audio):
USB with ardour 48K
* Playback Device 0 (USB Audio):
- Subdevice 0 (hw:QU24,0,0):
used by: jackd (PID 213602)
access: MMAP_INTERLEAVED
format: S32_LE
subformat: STD
channels: 32
rate: 48000 (48000/1)
period_size: 512
buffer_size: 1024
Is there any particular (technical or practical) reason why digital consoles would run only at a fixed rate of 48 KHz rather than allowing for sample rate changes like usual audio interfaces do?
Yes…
It is much easier to implement a single sample rate, especially on FPGA based consoles like the A&H consoles.
Given that the majority of live audio equipment, which the consoles in question are, runs at 48k, it means you don’t need to do sample rate conversions between different pieces of digital audio equipment, makes things much easier from an operational standpoint as well.
Many consoles that DO let you use higher sample rates are run at 48k because people won’t hear the difference, especially true in live situations where you have lots of other noise to contend with as well. This holds true as well of cheaper consoles that allow at least switching between 44.1 and 48k, most of the time they stay at 48k and never get switched.
It just makes no sense to dedicate extra processing power to higher sample rates when you can put in more features instead to run simultaneously.
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