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Try installing Ubuntu. I'm using 2i2 with a AT2030 through pulseaudio, and not having any issues in Audacity right now. My Ubuntu version is 16.04 with 4.4.0-13

Try installing Ubuntu. I'm using 2i2 with a AT2030 through pulseaudio, and not having any issues in Audacity right now. My Ubuntu version is 16.04 with 4.4.0-137 kernel. I remember having some crack/pop issues (you could literally see absent points in Audacity) with earlier versions of the distro/kernels, but I think they evaporated after upgrade/reinstall. So it's kinda unreliable/flaky, depends on some kernel config perameter maybe, or config file in the distro. I forced mine to work in 44100 sampling rate due to an unrelated hardware malfunction that causes loud pops during device initialization and sample rate switching(observable in windows also). In /etc/pulse/daemon.conf default-sample-rate = 44100 alternate-sample-rate = 44100 Don't know if it affects the recording pops/cracks. Probably not. Another issue I had that evaporated with updates is that pulseaudio/audacity recording used to work with ginormous latency, like a second or more, maybe caused by the crack/pop issue itself forcing pulseaudio to increase buffer size to absurd sizes or something. Another bit of sound configuration I found in my "how to reinstall ubuntu guide" I've written for myself: in /etc/pulse/default.pa add the following before 'load-module module-default-device-restore': load-module module-remap-source source_name=input1 master=alsa_input.usb-Focusrite_Scarlett_2i2_USB-00.analog-stereo master_channel_map=front-left channel_map=front-left format=s32le rate=96000 remix=no this line turns 2 scarlet's inputs interpreted as stereo by the os into a a mono source (probably requires customization of the device name)

nickjj 8y ago
Have you tried recording with ffmpeg (using pulse or even alsa), or even OBS studio? Those drop outs are exactly what I think is happening here, even though I never looked at the waveform in Audacity, that's exactly what it sounds like. If it happened mid-word, it was like that section of the word was removed from the face of the Earth. The problem is I'm using a very new kernel (4.19) hmm. I was using 44.1k too by the way (also tried 48k too) and set it exactly how you did in daemon.conf. Thanks, that is a handy line to have in case I can get all of this to work. Sure beats mixing it back to mono in post-production for every recording. Although isn't that resetting the rate back to 96000 when you want it to be 44100?
- nullifidian 8y ago
  >Have you tried recording with ffmpeg Just tried ffmpeg -f pulse -i default output.wav No issues observed. >Although isn't that resetting the rate back to 96000 when you want it to be 44100? Yeah, I just noticed that too. I inherited 96000 since before I resorted to using 44100, so... It just works, too lazy to change it. As for alsa without pulse, I tried using ardour with jack year or two ago, and no issues either. arecord --device=default --format=S32_LE -r44100 -c2 -t raw | aplay -r44100 -c2 --format=S32_LE --device=default - works too. So it's either software, or something else in your system is affecting it. As someone said in this thread, debugging such an issue could get very arcane.
  - nickjj 8y ago
    Could you try ffmpeg using aac instead of wave? I think it would involve adding only `-c:a aac` to the command. I'm curious now. Did you ever notice any type of pops and cracks during normal system usage? Not even recording. Just going to Youtube or any page that happens to interface with your sound card. pavucontrol was notoriously bad for triggering it. It sounds like you have a very clean connection.
    - nullifidian 8y ago
      >Could you try ffmpeg using aac instead of wave? I think it would involve adding only `-c:a aac` to the command. ffmpeg -f pulse -i default -c:a aac -strict -2 output2.m4a ffmpeg -f alsa -i default -c:a aac -strict -2 output2.m4a no issues. alsa one through pulse too of course. -strict -2 is because distro's ffmpeg is old. >Did you ever notice any type of pops and cracks during normal system usage? Not even recording. If I remember correctly, yes. When I had crack/pop issue playing youtube videos or music I would occasionally hear this. But again, it fixed itself. My general advice is to not use Debian unstable, and if you want a stable working enviroment avoid rolling release distros generally. When I tried living with Debian unstable 10+ years ago someting would constantly break on a apt-get upgrade. Not sure how it is now.
      - 8y ago

nickjj 8y ago

Have you tried recording with ffmpeg (using pulse or even alsa), or even OBS studio? Those drop outs are exactly what I think is happening here, even though I never looked at the waveform in Audacity, that's exactly what it sounds like. If it happened mid-word, it was like that section of the word was removed from the face of the Earth. The problem is I'm using a very new kernel (4.19) hmm. I was using 44.1k too by the way (also tried 48k too) and set it exactly how you did in daemon.conf. Thanks, that is a handy line to have in case I can get all of this to work. Sure beats mixing it back to mono in post-production for every recording. Although isn't that resetting the rate back to 96000 when you want it to be 44100?

nullifidian 8y ago
>Have you tried recording with ffmpeg Just tried ffmpeg -f pulse -i default output.wav No issues observed. >Although isn't that resetting the rate back to 96000 when you want it to be 44100? Yeah, I just noticed that too. I inherited 96000 since before I resorted to using 44100, so... It just works, too lazy to change it. As for alsa without pulse, I tried using ardour with jack year or two ago, and no issues either. arecord --device=default --format=S32_LE -r44100 -c2 -t raw | aplay -r44100 -c2 --format=S32_LE --device=default - works too. So it's either software, or something else in your system is affecting it. As someone said in this thread, debugging such an issue could get very arcane.
- nickjj 8y ago
  Could you try ffmpeg using aac instead of wave? I think it would involve adding only `-c:a aac` to the command. I'm curious now. Did you ever notice any type of pops and cracks during normal system usage? Not even recording. Just going to Youtube or any page that happens to interface with your sound card. pavucontrol was notoriously bad for triggering it. It sounds like you have a very clean connection.
  - nullifidian 8y ago
    >Could you try ffmpeg using aac instead of wave? I think it would involve adding only `-c:a aac` to the command. ffmpeg -f pulse -i default -c:a aac -strict -2 output2.m4a ffmpeg -f alsa -i default -c:a aac -strict -2 output2.m4a no issues. alsa one through pulse too of course. -strict -2 is because distro's ffmpeg is old. >Did you ever notice any type of pops and cracks during normal system usage? Not even recording. If I remember correctly, yes. When I had crack/pop issue playing youtube videos or music I would occasionally hear this. But again, it fixed itself. My general advice is to not use Debian unstable, and if you want a stable working enviroment avoid rolling release distros generally. When I tried living with Debian unstable 10+ years ago someting would constantly break on a apt-get upgrade. Not sure how it is now.
    - 8y ago