[HowTo] Rip Vinyl LP's in Manjaro

Difficulty: ★★☆☆☆

This tutorial will teach you how to rip Vinyl LPs in Manjaro but please know that hardware specific adjustments might be needed depending on your hardware.


  • Adjust alternate-sample-rate and avoid resampling in /etc/pulse/daemon.conf
  • use alsamixer to control your sound card,
  • record with arecord in the terminal.
  • Import/Edit the output file of arecord in Audacity.
  • Use Musicbrainz to add tags,
  • Enjoy!

The long version:

The Hardware used in this tutorial:

  • Technics SL-23 Record Player
  • AT-95ML cartridge and stylus
  • Darlington Labs MM-6 phono preamp
  • Drop SDAC USB 2.0
  • HP xw4600 Workstation with Realtek ALC262 (Creative X-Fi XtremeGamer)
    Sound card specs:
    – 24-bit Analog-to-Digital conversion of analog inputs at 96khz sample rate
    – 24-bit Digital-to-Analog conversion of digital sources at 96khz to analog 7.1 speaker output
    – 24-bit Digital-to-Analog conversion of stereo digital sources at 192khz to stereo output
    – 16-bit to 24-bit recording sample rates: 8, 11.025, 16, 11.05, 24, 32, 48, and 96khz

The Software used:

  • Manjaro Stable XFCE (and it’s base packages)
  • Musicbrainz Picard
  • Audacity (1:2.4.1-4 Official Repositories)
  • Terminal
  • ALSA (see below packages)
    • alsa-lib
    • alsa-utils
    • alsa-ucm-conf
    • alsa-topology-conf
    • alsa-plugins
    • alsa-firmware
    • alsa-oss


Manjaro stable uses PulseAudio to control audio devices and is a great solution for most users. However, as it is configured, it will resample audio above or below it’s default, both in capture and playback. When trying to capture bit perfect audio, it is best to bypass Pulse and record directly from the respective sound card through ALSA. For playback, it’s fine using Pulse audio when it was properly configured to playback without resampling, as it is simpler to configure Pulse so you can listen to the proper sample rate from any source (Audacious, Audacity, Google Chrome, Spotify, etc). Playback is covered in this section.


To record,

  • bring the RCA output of the phono preamp directly into the input of the sound card connected to the PC.

  • open a terminal and issue the alsamixer command.

  • From here press F2, then explore the options to show the devices for your system. Of interest are the options /proc/asound/cards, /proc/asound/devices and /proc/asound/pcm.

  • arecord will need to know the card and capture device in the format hw:card,device so make note for use later. For this tutorial, the correct card, device is found at hw:0,0 .

  • You can read more about ALSA Devices here

  • While still in ALSA mixer press F6 to select the sound card used for recording (In this tutorial HDA Intel is used as an example)

  • Press F4 to show the mixer for the capture devices for the selected card.

    • If you have multiple capture devices, ALSA mixer will have multiple controls for these capture devices too and fortunately, these are in order
    • To be safe, set this input to 5 to avoid overloading the card before finding out what level would be appropriate for recording.
    • Read more about ALSA mixer here.
  • Now that you have the correct input connected and identified in alsamixer, you will be able to prepare to record. Obviously a clean record, stylus etc, are essential.

  • To record, first prepare a list of commands that you can copy and paste to make life easier:

  • To monitor the recording rate of your sound card (hw:0,0 in this example, see above) use:

     watch -n1 cat /proc/asound/card0/pcm0c/sub0/hw_params

    you’ll need to adjust /cardX/pcmXc/subX/ for your input in ALSA identified above. For instance, for recording from the second sound card, first capture device (hw:1,0) instead of the internal card, use

    watch -n1 cat /proc/asound/card1/pcm0c/sub0/hw_params
  • Now to setup the correct command for arecord. Obviously adjust output directory, file name, recording format and rate to your liking. Optionally adding --duration=1500 will terminate recording after 25 minutes. This can be helpful for doing one LP side at a time, particularly if the turntable in use has an auto-return function and the recording is made unsupervised. For this tutorial, to record in stereo at 32bit 96khz to ~/Music/myfile.wav the command looked like:

    arecord --device=hw:0,0 --format=S32_LE --rate=96000 --channels=2 --vumeter=stereo ~/Music/myfile.wav

    Note: you may have noticed the sound card in this tutorial doesn’t list support for 32 bit capture, but when the command to record in s24le was issued, it stated the options were s16le or s32le. Obviously it is best to opt for the higher bit depth. It has been have confirmed by another user on another distribution that this same thing has been experienced. Perhaps an ALSA specific bug.)

  • With the correct commands ready, open two terminal windows (in addition to the already open alsamixer window).

  • In one terminal issue the command to monitor the capture rate of the soundcard (which will show closed until recording begins) and the arecord command above in a separate terminal to start recording.

  • Place the needle on the record then monitor the arecord VU meter and adjust the alsamixer capture (select Capture and use up/down arrows) until it is peaking around 50%. For the LP’s used in this tuturial this was somewhere between 8-10.

  • Once the correct level is set in alsamixer, reset the needle at the beginning of the record. Alternatively, optionally press ctrl C to stop the recording, delete the recorded file in the location specified in the arecord command, restart the recording in arecord, then reset the needle at the beginning.

    Here is a look at the sample workspace during recording:

  • Once both sides of the album have been brought in, issue CTRL C in arecord to terminate the recording, then move on to import/editing.

  • NOTE. There are instances where recording times may be long or require more drive space (Double LP’s, higher sample rates, unattended recording). arecord will automatically split any recording above 2.1 gb into a different file, which could require re-recording if in the middle of a track. Best practice is to monitor recording, eliminate long pauses without information, and split double LP’s into separate recording (one .wav file/record).

Playback and Editing

  • To listen and edit the file which was just recorded we need to make some simple changes to Pulse Audio to allow playback without resampling.

  • First open /etc/pulse/daemon.conf in a text editor

     sudo mousepad /etc/pulse/daemon.conf

    and find and edit (the edited lines must be uncommented in daemon.conf by removing the semicolon)

    ;avoid-resampling = false 


    avoid-resampling = true


    ;alternate-sample-rate = 48000`


    alternate-sample-rate = 96000`
  • Match the alternate-sample-rate to the same sample rate used to record in arecord above. Some prefer to rip at 192khz, in which case this would be set in the alternate-sample-rate = 192000. For this tutorial, it is set to 96000.

  • Once those changes are saved, issue

    systemctl --user restart pulseaudio

    to restart PulseAudio with the changes.

  • Now, to check the changes have stuck, find the file just recorded i.e. ~/Music/myfile.wav and play it with an audio player.

  • While the file is playing issue

     pacmd list-sink-inputs 

    Check the correct bit depth and sample rate are shown, and importantly the resample method shows copy or none, as seen below:

    [david@david-protools ~]$ pacmd list-sink-inputs
    1 sink input(s) available.
      index: 1
      driver: <protocol-native.c>
      state: RUNNING
      sink: 0 <alsa_output.usb-Grace_Design_SDAC-B_USB_2.0-00.analog-stereo>
      volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
              balance 0.00
      muted: no
      current latency: 494.39 ms
      requested latency: 460.00 ms
      *sample spec: float32le 2ch 96000Hz ##input sample rate is correct*
      channel map: front-left,front-right
      *resample method: copy ## no resampling taking place*
      module: 14
      client: 5 <Audacious>
      	media.name = "Audacious"
      	application.name = "Audacious"
      	native-protocol.peer = "UNIX socket client"
      	native-protocol.version = "35"
      	application.process.id = "46445"
      	application.process.user = "david"
      	application.process.host = "david-protools"
      	application.process.binary = "audacious"
      	application.language = "en_US.utf8"
      	window.x11.display = ":0"
      	application.process.machine_id = "675df23d99b8436bb7b9875ef0b74bd4"
      	application.process.session_id = "2"
      	application.icon_name = "audacious"
      	module-stream-restore.id = "sink-input-by-application-name:Audacious"
  • This can also be confirmed by monitoring the output of the USB DAC (hw:2,0) in this tutorial by issuing

    watch -n1 cat /proc/asound/card2/pcm0p/sub0/hw_params

    Where card2/pcm0p/sub0 represented the playback output of my USB DAC in ALSA. Obviously adjust to your hardware.

    IF you’re showing 44100 sample rate be sure Pulse Volume control is not currently open when checking with above commands. If it is open it will lock the default sample rate defined in daemon.conf file


Now Pulse is properly configured to playback without resampling, import the recorded file into Audacity to begin editing.

  • Launch Audacity (a separate workspace can be used to keep things tidy)

  • Before importing the file into Audacity some setup is needed to edit at the same bit depth and sample rate as the source material.

  • Navigate to Edit>Preferences and adjust as needed for the recorded file. For this tutorial set to 32bit and 96000hz.

    Since the source was in the same bit depth and sample rate, and the output hardware can handle these rates, no resampling should occur. Not a huge deal here, since we are just using this to edit the master file into separate songs and add some basic tags for Musicbrainz to use later.

  • Now go to File>Import>Audio and select the file recorded from the directory used to output from arecord earlier. i.e .(~/Music/myfile.wav)

    If done properly, it should import to a single track and all the waveforms should be well below the max 1.0 of the meter on the left hand side of the track. If the waveforms look like a block, the audio was recorded too loud and will add distortion. It is best to have Audacity configured to show clipping (View>Show Clipping) and it can be seen in the below example from this tuturial, where the needle drop on side B registered as red line. You DO NOT want to be seeing these red lines within the waveforms, as the audio has been clipped during recording and will need to be re-recorded with a lower capture volume.

  • Next, familiarize yourself with the slip tool (EDIT>CLIP BOUNDARIES>SPLIT) and set a keyboard shortcut for Split under (EDIT>PREFERENCES>KEYBOARD).

  • For more information on Audacity Split editing check here

  • Edit the master imported file to remove dead space at the beginning of the record prior to music start, in between record sides, and at the end after the music has stopped. Ensure all needle drops/clipping are removed.

  • Use CTRL A to select all, then CTRL J to join the sections together.

  • Go through the entire file to isolate clicks and pops and remove them with the pencil tool. Pay extra attention to every tall spike to ensure they are not clicks and pops. See below for examples:

    Right Channel Pop


    Zoomed for correction

    Post Correction

  • With the entire file now edited and joined, normalization will be used to raise the volume while preserving the song and album dynamics. Choose EFFECT>Normalization and set as below, or to a peak amplitude of your choice. Generally not recommended to go above -0.3 dB.


  • Avoid normalizing channels independently as a pop or crack, or single channel instrument may prevent proper normalization and you’ll get an overloud channel!

  • Avoid using the Effects in Audacity (noise removal, click removal) apart from normalize and the pencil tool to edit out loud pops manually as described above. More about Audacity Effects can be found here

  • The last step before export is to isolate each song. Find the beginning of each track and use the split function to separate. With exception of the first and last song, the end of one track should be the beginning of the next. Continue until all tracks are cut.

  • To Export, highlight each song individually, and use FILE>EXPORT SELECTED AUDIO. Choose FLAC from the bottom right, Level 8 and 24 bit. This returns a 24 bit 96khz flac file. Export to a Folder with the album name to make life easier in MusicBrainz.

  • Fill in the appropriate Tag information on the next window, and move on until they’re all done! (If you’re not getting Metadata information at export check EDIT>Preference>Import/Export>Show Metadata Tags Editor Before Export is checked)


Once all the tracks are edited and exported into the correct folder-

  • Open Musicbrainz Picard , choose Add Folder and select the folder with the .flac exports from Audacity.

  • Once the files are in, select Unclustered Files, then press Lookup.

  • Let Musicbrainz do it’s magic and save the new tags.

  • Optionally install a plugin for playlist generator within MusicBrainz and right click on the album, plugins, generate playlist.

More can be found about Musicbrainz Picard here.


Navigate to your new rip, double click on the playlist and Enjoy! Feel free to run the commands up above to verify you’re getting the correct playback bit depth and sample rate. FWIW, my USB DAC only outputs in 32bit, so my 24bit rips, show playback at 32 bit depth when I playback on my PC. My main reason for ripping my vinyl collection is to listen to them upstairs on my good stereo, which can output 24bit 96khz without issue.

Note 1: Please feel free to share any improvements in the thread below.
Note 2: Please do not turn this thread into a discussion of the merits of digital vs analog, audibility of sample rate etc, etc: this tutorial is all about how to properly record records at a high bit depth and sample rate in Manjaro Stable for those who want to. If you’re happy with your streams or CD’s, more to you, this wouldn’t be the conversation for you. :bowing_man:


This is a fine piece of work!!

Great job!


This is a mighty fine report, and thorough

One change I would suggest is to replace command pulseaudio -k and use systemctl --user restart pulseaudio

The -k or --kill option will change the pulseaudio.service to inactive(Dead) and make other changes to the PulseAudio that are not usually visible (pasystray is the only GUI that can show the changes)

I would also suggest just for appearance to use a simple list of the core ALSA packages instead of a screenshot

  • alsa-lib
  • alsa-utils
  • alsa-ucm-conf
  • alsa-topology-conf
  • alsa-plugins
  • alsa-firmware
  • alsa-oss

alsa-card-profiles is not an ALSA package and not relevant for this

I have some other ideas that might improve workflow or reduce changes to the audio data
But critiquing a dude’s mix can get contentious, so you might want to consider :

  • Stop PulseAudio (and/or Pipewire) and do it all in ALSA (no resampling possible)
  • IF you get a few mis-takes in recording level, record direct into Audacity to see the audio levels and, if necessary, abort the recording
  • Don’t use normalization. Export the recording to FLAC as is and use metaflac --add-replay-gain to add gain adjustment metatags (no change to the audio data)
    FLAC - documentation

Wow, thanks for the contribution. I made the changes to the alsa list and restarting pulse instead of killing it, as you suggested.

Your other changes are noted as well.

I explored stopping Pulseaudio completely to record, but opted to just modify Pulse to play nice with the ripped files instead. This left the default Manjaro audio pipeline intact for applications that did not play nice with ALSA.

Problem with recording into Audacity is it was difficult for me to decouple it from Pulse completely (without disabling Pulse completely). I even overloaded my audio card once playing around with pavucontrol and alsamixer : \ and had to pull the pc power and discharge the mobo to get it back up!

Which leads me into the bit about normalization

From the research (here) I did it seems the majority consensus was to capture at a lower volume, and use normalization to bring the recording to an appropriate volume for playback. Thus my recommendation in the tutorial to start at 5 and slowly work up, then normalize to bring to playback volume.

Now normalization has it’s nuance, but the albums I’ve ripped seem to have their dynamics maintained (both in song and album).

Replay Gain is a great thing, but not all players support it, and since I didn’t have much knowledge of implementing it, didn’t include anything about it.

However, it’s definitely something I’ll explore.

Thanks again for the tips!


Thank you for your consideration

I learned about replaygain from working in radio, but most other audio professionals use some form of normalization.

With your workflow there would no benefit from scanning for ‘album’ gain because normalizing the entire album will keep the audio level consistent
For a playlist compiled from multiple albums, a ‘track’ gain adjustment might help to adjust for differences in levels between albums, but that is unlikely to be needed if the final normalized level of all .flac files are kept within a narrow peak level range

I agree with Steve Hoffman that maintaining (or in his case restoring) the full dynamic range of an audio recording is essential to keep the original spirit of the music alive


You know I went on a deep dive of methods folks use, including Replay Gain method, which is recommended by AES for high fidelity audio capture.

However, based on the Library of Congress analog media archiving white paper, and my own subjective review, I began utilizing the peak capture method. While more work, I believe it captures the analog signal as digitally transparent as possible.

That said, many over on the discussion I started on Audio Science Review indicated whole album normalization is a tried and true method which is digitally transparent when done correctly. Thus I’ll leave it in the guide above since it simplifies the process and the guide.

One good thing, is folks can do their research and choose for themselves how they’d like to do things!