Unwanted distortion in Reaper

[IllustratorPascal]

Hellol,
I have installed Reaper on my computer and it works.
The system is as follows
- Linux Mint 18.3 Cinnamon 64-bit
- AMD Ryzen 7 - 1700X eight core
- 32 GB ram
- Nvidia graphic card 1060

I have my external audio mixer connected to a Behringer U-control UCA 202 audio capture device that convert analog signals to usb.
The Behringer U-control also has an optical output that is connected to the optical input of my active speaker system (Logitach Z5500).

I started Jack with the QJack application.

When I produce sounds with Reaper it sounds very distorted in Linux. I also have Reaper for Windows on the same computer (dual boot) and there everything sounds fine)

The distortion in Linux is for:
- software instruments
- external hardware instruments
- recordings that I drag and drop in Reaper
So actually for everything

I don't have added a distortion plugin so that's not the case.
There also nothing clipping. Levels in the mixer looks fine.
The sound of the distortion sounds a little bit like there's a lo-fi plug in added.

Does anyone have an idea what this can be? I can't figure it out.

[lilith]

Please post your Jack settings or make a screenshot, i.e. sample rate, buffer size, periods / buffer. What happens when you switch to ALSA instead of Jack in Reaper? Do you get good sound from other applications?

[IllustratorPascal]

In the attachment the Jack screenshot.
When I choose ALSA in Reaper I have an error message - see attachment - and no sound.

[lilith]

Try setting the buffer size (= frames/period) to 1024 first and check if the sound changes. You have to stop and restart Jack after the change. If Reaper can't use ALSA maybe your soundcard is occupied by an other application. Do you use PulseAudio also?

Is anything written in the log window of Jack?

[IllustratorPascal]

Yes! It works. I changed the buffer size from 64 to 1024 and then the sound is fine like it shoud be
I don't know what PulseAudio is, but when Jack is started I noticed that a Youtube video stopped working. When I stop Jack the video works again.
Luckily I don't need to watch video's when I record audio in Reaper.

[lilith]

Great, if you need less latency you can go down again to 512 and 256 until you run into problems like dropouts (xruns). With bridged pulseaudio or bridged alsa it's possible to have sound from Reaper and e.g. youtube at the same time. So, in case you need this it's possible with some tweaking.

[IllustratorPascal]

Thank you very much.

[bluebell]

Try setting the buffer size (= frames/period) to 1024 first and check if the sound changes. You have to stop and restart Jack after the change.

Jackd's buffer size can be changed on the fly, see http://marzen.de/Holger/jack_bufsize/

But some client programs have to be restarted after a change.

[khz]

Code: Select all

LM@LAW ~ $ jack_bufsize 1024

96/128/256/512/1024/2048/4096

[AlexTheBassist]

Great, if you need less latency you can go down again to 512 and 256 until you run into problems like dropouts (xruns).

This is unplayable, because a total roundtrip in this case will be above 30 ms, which is noticed even by non-musicians. However, good sound cards like Presonus ones can be playable with 256 samples/buffer also, but only in case USB works fine with 2 periods per capture.

[AlexTheBassist]

Code: Select all

LM@LAW ~ $ jack_bufsize 1024

96/128/256/512/1024/2048/4096

96 is unsupported by some convolution software, like JConvolver, which can be found in Guitarix. It's better to use power of 2 buffer sizes to get more stable performance.

[lilith]

Great, if you need less latency you can go down again to 512 and 256 until you run into problems like dropouts (xruns).

This is unplayable, because a total roundtrip in this case will be above 30 ms, which is noticed even by non-musicians. However, good sound cards like Presonus ones can be playable with 256 samples/buffer also, but only in case USB works fine with 2 periods per capture.

For me everything <= 10ms is ok.

[AlexTheBassist]

For me everything <= 10ms is ok.

A total roundtrip is always longer than samplerate / (buffersize * n). There's an additional delay in the soundcard itself (that's why I love Presonus, they add up only 1-2 ms), there's a transfer delay on USB bus. To get to real 10< ms, you need to use buffer size of 64 or even 32, depending on your hardware.