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Audient ID48 - Initial Transient Distortion

jumper981

Member
Joined
Jan 1, 2020
Messages
45
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Hi to everyone!
I recently bought an Audient ID48.
I found that in some conditions, the outputs will distort/clip on initial transient.
DAC outputs should have CS43198 despite Audient says “Sabre” on his site.

I’ve conducted some measurements and attached there.


I reproduced this issue in this way:

Ableton Live
Open a MIDI Channel with a good low frequency sine wave with VST (Serum 2 used here)
Attack Parameter relatively fast and sine wave phase starting from triggering.
Filter it with a steep low pass filter to clearly leave only the sine wave.
Do some fast triggering and you will find the distortion on triggering only.

Other method:
I’ve downloaded a high quality 36Hz sine wave with phase starting at 0deg.
Do some fade at both starting and ending.
Filter it same as before.
Distortion only on fade portion.
 

Attachments

In your attachment you have:
  • Line OUT 1+2 into 1+2 Line Input (Sine Wave Download, Fade): Noise dominates over an extremely quiet 36Hz sine. See below
  • Sine Wave Fade: There is literally no signal. Only 0s.
1.png


I suspect what's happening is that you are causing clipping by playing files with an abrupt start or by triggering the files on/off. There is nothing wrong with your interface.

DAC outputs should have CS43198 despite Audient says “Sabre” on his site.
This doesn't matter.
 
Hi, thanks for your response.
I checked the files personally and left near 10mS start if I remember correctly.

All my other interfaces and DACs doesn’t exhibit this behavior with the same files.

I quoted the Cirrus Logic chips cause the distortion vs output level of this chip could be strange as posted here in a research from a user in AudioScienceReview.

I suspect something from the CS chip or the audio asio driver of the interface.
 
I noticed it while making music.
if I’m synthesizing a kick drum it make absolutely sense to use a sine wave in this way.
Despite this, how I use it it’s not an argument of this post.
 
As I said before, original audio playback have the right fades.
Also, those were starting at phase 0, so it’s not even necessary.
 
I played back the first file very loudly and heard clipping. When looking at the waveform in the spectrogram I see a lot of spectral noise exceeding a very low-level sine, which means: clipping.

The second file contains literally nothing. No audio data.
File Summary
File Name : Sine_Wave_Fade.wav
sox WARN wav: wave header missing extended part of fmt chunk
Channels : 2
Sample Rate : 44100
Precision : 24-bit
Duration : 00:00:00.68 = 29975 samples = 50.9779 CDDA sectors
File Size : 240k
Bit Rate : 2.82M
Sample Encoding: 32-bit Floating Point PCM

File Detail
sox WARN wav: wave header missing extended part of fmt chunk
Samples read: 59950
Length (seconds): 0.679705
Scaled by: 2147483647.0
Maximum amplitude: 0.000000
Minimum amplitude: 0.000000
Midline amplitude: 0.000000
Mean norm: 0.000000
Mean amplitude: 0.000000
RMS amplitude: 0.000000
Maximum delta: 0.000000
Minimum delta: 0.000000
Mean delta: 0.000000
RMS delta: 0.000000

It is a problem to do with the digital domain, not your gear. How you are creating and processing the signal.
 
Those file should be the recorded in/out of the interface. I think that I didn’t include the original sine wave.
 
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