#Linux Audio

[daemonhxckergrrl]

pipewire my beloved my beloathed, please, please share your secrets. i just wanna do a little automatic routing and ditch a bunch of mix controls i don't need

[cracklestatic]

A Workflow for Podcast Audio in Audacity

For some time, I have been using this workflow for processing raw audio into clear, clean, production quality files for ebooks and podcasts.

These steps are for use in Audacity; you can adjust them as necessary for other editors.

Original Recording Quality & Format

First, make sure the original is recorded in as high a quality as possible. Follow the easy to find best practices: quiet location, quality microphone, wind screen and pop filter on the mic, etc.

Make it a high priority to capture the voice (or voices) at a moderate level, but never drive the recorder into clipping. Make sure there is as little hum as possible on the audio. If necessary, use battery power for the mic and recorder. If you have problems with power line hum, see the tutorial for reducing power mains hum in your audio chain

As to format, you cannot go wrong with 24 bit / 96 kHz sampling, if your equipment can do it. Otherwise, use 16 bits if you cannot record at 24 and 48 kHz sampling if you cannot run at 96 kHz.

I know some older, legacy artists may still use CD sampling at 16 bits / 44.1 kHz. Unless you are making CDs, don't bother with that. The future is sampling in multiples of 48 kHz.

1: Sync if Multiple Tracks

When I get the raw recordings, the first thing I do is open a new project in Audacity and - if there are multiple participants - get all of the tracks synced in time. Syncing isn't necessary if everyone's mic is recorded on one multitrack device.

Line up the tracks on some kind of identifiable sound as early as possible. Use a hand clap, count-in, or something you can see on an audio envelope or spectrogram.

If there are multiple recorders in use, you'll notice that the other tracks eventually drift out of sync. Make the best track the "main" or "master" and then stretch or shrink the other tracks to match.

All of the text below refers to one track, but should be done for all tracks it you are working on a podcast with multiple tracks.

Leave a few seconds of dead air / room tone, containing the ambient background sounds, before and after the program.

2: Normalize the Audio

Normalize the sound level of all of tracks to -3 dB relative to full scale (-3 dBFS).

3: Cut Out the Cruft

Cut out sections with extended silences. Make the same cut across all tracks, as it is essential to keep the whole project in sync.

If there are sneezes, coughs, or odd background noises on any track, replace them with silence.

4: Apply Noise Reduction

Apply noise reduction to tracks individually. Audacity is great for identifying and removing rhythmic sounds, such as fans, motors, or anything which whines or buzzes.

To get a profile for the noise, find a few seconds with none of the desired content - some dead air between sentences, for example. Highlight it and run the noise reduction "get profile" function.

After profiling the noise, highlight the whole track and apply noise reduction. Use the maximum number of frequency bands. Don't attack the noise too aggressively, as it makes the recording sound hollow and unnatural. Just apply 6 dB of noise reduction.

If the noise is still there, take a new profile then apply another 6 dB of noise reduction.

5: Re-Normalize the Levels

Normalize the sound level of all of tracks to -3 dB relative to full scale (-3 dBFS).

6: Apply the Noise Gate

The noise gate lowers the background level when there is no strong signal, as between sentences. Work with the settings. I like to set the attenuation to about - 20 dB, with only a few milliseconds (ms) of attack time, about a second (1000 ms) of hold time, and a half second (500 ms) of decay. But work with what you have and go with what sounds best for your situation.

7: Re-Normalize the Levels

Normalize the sound level of all of tracks to -3 dB relative to full scale (-3 dBFS).

8: Apply Amplitude Compression

To bring the peak and average voice levels closer, which is to make the sound more even, use the compressor. Depending on the plugins you have, there may be several choices available. There might be more settings or fewer settings to tweak. In a general sense, these settings work:

Look ahead time: 10 ms

Threshold: -26 dB

Knee: 3 dB

Ratio: between 3:1 and 5:1

Attack: 5 ms

Hang or Hold: 200 ms

Decay: 300 ms

Adjust the threshold, compression ratio, and timings as necessary for best sound of your content.

9: Apply Limiting to Catch the Strongest Peaks

The compressor will let some of the loudest peaks get through, although they will be moderated. Use the limiter to catch those peaks.

Look ahead: 2 ms

Threshold: Set to match level of other peaks.

Release: 30 ms

10: Re-Normalize the Levels

Normalize the sound level of all of tracks to -3 dB relative to full scale (-3 dBFS).

11: Amplify / Attenuate for Platform Compliance

Amplify the whole project + / - a few dB as necessary to meet the requirements of the distribution platform. Whether Spotify, YouTube, SoundCloud, or other platforms, there is some kind of standard for peak and average loudness levels. Also, there is often a requirement that noise be below a certain level.

12: Set the Pre and Post Empty Space

Trim off the dead air before and after the program content. I find that one second of room tone before the start and four seconds after the end is sufficient most of the time.

[catbirdlinux]

What will you create today? Make it on a worthy operating system. Catbird Linux is the real deal - power and freedom for your PC.

[natural-blogarithm]

windows and OSX should be banned worldwide so that everyone has to switch to a linux-based OS and people finally start making programs that run on it

[chahleybros]

Switching to Linux 🐧

Over the past couple years Windows 10 has been giving me progressively more asinine problems to deal with, from desktop issues and bloatware, to automatically installing out of date drivers to replace the ones I've manually installed.

Before you ask, I attempted using Windows 11 when it came out and I refuse to acknowledge its existence after the hell it put me through.

Most recently Windows 10 has seemingly made a point of making ASIO and all audio devices and software I use functionally useless, causing ridiculous amounts of crackling audio. No rollbacks fixed it, no re-installs fixed it, not even Reloading Windows fixed it. The reason for using ASIO is that Windows default audio protocol, WDM, has too much latency for live real-time use. This issue not only makes things like live-streaming basically impossible but it also outright made my Digital Audio Workstation and Video Editing Software incapable of processing audio for longer than a second without hanging. Both Mixcraft and DaVinci Resolve became useless on Windows 10, and OBS was barely holding it together. When I tell you I did everything to resolve this, I mean I spent an entire Month troubleshooting this. The only solution I could find was to abandon Windows all together.

I've been meaning to switch my Setup to Mint Cinnamon for a while as I've been testing it out in Virtual box for a year or so, and after this nonsense I got a new NMVe basically immediately to run Mint on, and after switching properly I have to say I'm likely to never use Windows as my Host OS again. And yes I tested Windows on the NMVe. The issue persisted there aswell. And testing Mint via Virtual Box on Windows with Hardware Exposed via PAE and AMD-V showed the issue was entirely on Windows.

For reference this is my current rig: Everything is the same as when using Windows 10

GPU: Nvidia RTX a4500 20gb

CPU: AMD Ryzen 5 5600g

Ram: 16GB 3200mhz

DAC: PreSonus Audiobox iTwo

Capture Card: AverMedia HD Mini GC311

The Results:

Not only does Mint Cinnamon not have any of the issues I've had with Windows over the past Decade, which still persist to this day, but literally every single device I have just works.

Basically everything is Plug-n-Play with <20ms of latency on the Capture Card, which on Windows had a minimum of 600ms delay using drivers required for the card to even be detected. My Audio interface also works even better than before, with neither my DAC or Capture Card requiring manual driver installs, as their protocols are Native to the Linux Kernel.

Proton lets you play any Windows game on Steam that doesn't have explicit Linux Support, with Wine available for desktop applications. Both run better than a Native Windows install due to Mints lack of Bloatware.

Windows 10 has so much built-in Overhead that Mint can Emulate it through Virtual Box faster than if Win10 were the Host OS.

In Short: Fuck Windows. :) Also this is how my desktop looks currently:

You can make custom start menus. :)

[andmaybegayer]

Should I be using pipewire over pulseaudio and if so why?

If you're on Wayland you should definitely at least check out Pipewire, because Pipewire and Wayland both follow a similar philosophy and are kind of being developed side by side. Pipewire with portals is one of the ways to do screen captures and screenshots on Wayland.

If you're running Flatpaks then Pipewire's integration with Flatpak Portals and xdg-desktop-portals more generally will simplify media handling for Flatpaks and generally make running Flatpak media applications more reliable and seamless.

If you use Bluetooth audio, Pipewire has simpler first class support for a wider array of Bluetooth codecs (high bitrate SBC/AptX/LDAC/AAC) and generally simplifies the process of setting up Bluetooth devices exactly the way you want over Pulse.

If you currently fight with running Jack sometimes (or worse, simultaneously running Jack and Pulseaudio) then you should definitely check out Pipewire, because Pipewire implements both Pulse and Jack compatibility layers that are way easier to look after and which can run simultaneously without any fuss.

If you're doing music production with Pulse, Pipewire's pro audio mode might give you some small quality of life improvements by reducing latency and improving inter-program audio links.

If you're doing a lot of live video stuff, especially video involving desktop capture on Wayland, Pipewire can simplify shuttling video around because in addition to handling audio, it handles arbitrary media streams, but you might have this worked out however you're already doing it.

If you are just running a standard X11 desktop and have no problems using Pulseaudio right now, you probably won't notice any change if you switch to Pipewire, especially if you aren't running Flatpaks or Bluetooth. Since Pipewire currently implements a lot of stuff through a Pulseaudio compatible interface, your normal actions with pactl and pavucontrol will continue working transparently or with minimal changes if you do switch.

Installing Pipewire is relatively easy if you don't have any custom pulse configuration. You just have to remove Pulse and install the pipewire, pipewire-alsa and pipewire-pulse packages.

[amalgamasreal]

FOSS IS LOVE

FOSS IS LIFE

[sineherepls]

shout out linux because if i was doing what i was doing on windows it would have crashed like 2 minutes after pressing the spacebar

also reaper is just awesome anyways

i used to use logic pro for a while but that's only because the college im going to demanded i used it for the first year but im free now... my shackles have unbound and now i can use reaper

also i do not recommend hackintoshes for real work lol i cant afford a macbook but it served me quite well for the year so i have no complaints besides it being significantly slower than on macbook hardware and also that stupid annoying audio issue i had with my sound card due to incompatibility so i had to use a usb DAC whenever i wasn't plugged in to any interface

also i do wish that ableton was on linux otherwise i would be using that with max/msp... and im not even going to bother using wine for it although i'll take a look at it someday

for now, reaper is awesome and yabridge is a perfect combination for all of my totally legally purchased dmgaudio and fabfilter plugins

[moonsinkfoxgirl]

a girl and her runestone, this too is yuri

[s-lycopersicum]

Coming to terms with the cruel reality that music sounds better through my phone.

[daemonhxckergrrl]

anyone messed w/ pipewire routing ? specifically automatically routing to different sinks based on an application's type.

stuff like discord can be done from the application side, but how would i tell eg all music players to route to a media sink, or web browsers to an applications sink /

[allylikes]

Recently I discovered MPD (Music Player Daemon), a music player based on a server/client architecture.

The design allows it to be used with a variety of different clients, all at the same time. That means you can interface with the server through simple command-line commands (mpc), through a terminal user interface (ncmpcpp), through a graphical user interface (ario), or even a web interface (ympd).

Setting it up isn't too hard, and the example configuration file (usually found at /usr/share/doc/mpd/mpdconf.example) provides lots of information and documentation about the available options. One thing you should keep in mind is that MPD's systemd service will use the config file at /etc/mpd.conf. I don't use the systemd service, so I launch it with the command $ mpd ~/.config/mpd/mpd.conf.

Once you have your server set up, you need to make it scan your music directory. All clients should have this functionality, but the most simple way to do it is with the mpc client, running $ mpc update. You should only need to do this on the first startup of MPD, and whenever you make changes to your music directory. Confirm that your library has been loaded with $ mpc stats.

To integrate MPD into my workflow, I wanted to implement keybinds to control my media. I know it can be integrated into playerctl (with mpDris2), but I would like separate media keybinding for playerctl. With that in mind, I chose to make my MPD keybinding use the super key + the media keys. In order to do that, I made 3 scripts: mpd-playpause, mpd-volume, and mpd-next. I used DWM's config to bind the relevant keys to these scripts.

These scripts (along with explanations and screenshots) can be found at https://github.com/allylikesu/mpd-scripts.

In the future, I want to implement a little dmenu control panel for MPD , so watch out for that!

[catbirdlinux]

Catbird Linux v3.2 is USB pluggable Live Linux for writers, programmers, data crunchers, and content creators.

[dibelonious]

Easyeffects - the best Real-Time Audio Processing.

[bluastro-yellow]

we had to ditch our old broken dumb tv and buy a smart one and I'm losing my mind because apparently you can't fuck with webOs that much. like you can install things like kodi but whatever, but you have to root it(?) which I'm not risking because it voids any warranty

it doesn't even have vlc media player

[lensdeer]

I hate modern GTK, but Helvum is SO MUCH NICER TO USE than qpwgraph jfc

Why wouldn't you group sinks with their monitors aaaaa it makes no fucking sense and needlessly complicates everything