I'm running Ubuntu 24.10. My /media directory (the usual mountpoint for external HDD) has the following permissions: $ ls -l / | grep media drwxr-xr-x 4 root root 4096 mag 16 2024 media The owner is root but it has permissions to execute (aka enter) the directory. From normal user I can do: $ ls /media/ mark root if I run Audacity as normal user (installed from snap) and try to open an audio file it refuses to enter the /media directory: > Error opening directory /media: Permission denied I tried: sudo chmod -R a+rxw /media/ but nothing has changed. What should I do in order to let Audacity entering this directory?

I am on Fedora 25 (x86_64) and I use XFCE desktop. I need to record a sound in sync with a video. Recording is done by Audacity and video playback is done by VLC. Currently, I have set global keyboard shortcut for VLC to pause and play the video when I want. I use another keyboard shortcut in Audacity (not global) to append my voice to the sound being recorded and I hit space (local) to pause the recording. You see that this makes the recording asynchronous to the video playback, which sucks! My question is how can I define global keyboard shortcuts for my desktop (currently XFCE but I can use other desktop environments on my Fedora if necessary), to start playback and record at the same time (with a single key [combination] and pause them so (with a single key [combination]) as well? To make the question clearer, for example I have set CTRL+ALT+SPACE as global hotkey for VLC for play/pause. How can I define the same key combination for Audacity to do Append Record/Pause when I send this key combination to the system?

I'm listening to some recording of a talk, which was made with a microphone far from the speaker; there's some noise and perhaps a bit of echo and it's not super clear. Now, if this were just an audio file on my PC, I would (try to) use Audacity which is supposed to be able to help. But - this is a video clip played by some web page on my browser, and I can't (or find it difficult to) download it. Is there a way I could apply some kind of filtering on my outgoing audio to clear it up? My system: Devuan GNU/Linux Excalibur (i.e. Debian Trixie without systemd) on an x86_64, with Cinnamon as the desktop environment. Sound server is pulseaudio.

*I guess this is sort-of a software recommendation question, but if I post it there I think even less people will be familiar with this specific setup (5.1 surround, Debian 12, KDE and PipeWire)* I recently upgraded from PulseAudio to PipeWire which enabled 5.1 on my setup surprisingly. I got a USB SoundBlaster that feeds *Digital Surround 5.1 (IEC958/AC3)* (compressed 6-channel audio through TOSLINK optical) into my sound system. For the purposes of the question, the OS sees this as 6 individual channels, and I'm able to play for example multi-channel FLAC files in VLC, or even Firefox. The built-in KDE sound applet in the System Tray as well as the System Settings > Sound allow me to individually control the volume of each sound channel, however only the applet shows a VU meter and it's in mono/stereo. **Is there a program / plugin that can show the individual volumes of each sound channel, sort of like Audacity has its "Mixer Board" which displays the VU level of each individual track.** I would be OK with any full fledged Gtk/Qt app, a KDE add-on, even a cli terminal app that shows the VU levels. Less preferable but even VLC add-on would at least do the job. I recently started making multitrack mixes of different songs, and my own music I play on guitar into 5.1 mixes, but they are hard to balance just by listening, I want something visual as it plays. (Audacity shows these visually, but cannot output 5.1 audio during the mixing, only after it has been exported to FLAC and played with a different app like VLC). I tried Easy Effects which is an amazing new app made for PipeWire, however they do not support multi-channel and do not plan on doing it as it would require a total rewrite (according to their Git/issues history).

I'm facing the issue that.. $ audacity ..starts audacity, but with missing inputs. $ pasuspender -- audacity ..starts audacity with all my inputs restored, so I can record from line-in, but then I cannot listen to music while doing so and there's some other strange behavior when trying to playback what I recorded. In some parts the audio gets extremely loud and then back to normal. So it's not even a viable workaround. All of this worked without problems on the same hardware before I upgrade from Linux Mint 19 to Linux Mint 20 Audacity v2.3.3 $ lsb_release -a No LSB modules are available. Distributor ID: Linuxmint Description: Linux Mint 20.3 Release: 20.3 Codename: una $ cat /etc/upstream-release/lsb-release DISTRIB_ID=Ubuntu DISTRIB_RELEASE=20.04 DISTRIB_CODENAME=focal DISTRIB_DESCRIPTION="Ubuntu Focal Fossa"

I want to install the GVST plugin pack so that Audacity (from Flatpak) can see it. I am using KDE Neon (fairly new, I'm not sure how to find the version number). I've tried various directories to copy the .so plugins to, but so far no success. I have restarted Audacity each try to make sure it refreshed everything. If possible, I'd like other VST supporting Flatpak apps to see the plugins also.

Every time I'd like to update my OpenSuSE Leap 15.3 (x86_64) system, the software updater in the systray stumbles upon a recommended update for audacity (2022-55 (1)). It *always* fails to get installed due to a missing ffmpeg-3 package: > nothing provides 'ffmpeg-3' needed by the to be installed audacity-2.2.2-bp153.3.3.1.x86_64. What's wrong here? An unresolvable dependency in a distro is somehow strange, isn't it? **UPDATE:** These are the repos configured in my YaST:

I'm trying to install some synths on audacity. I'm running PopOS 20.14. The audacity documentation says you should install plugins to /Library/Application Support/audacity. I cannot find this folder on my machine though. I did find an audacity folder in the lib folder but that has no plugin folder so I'm again wondering if this is where I should be pasting the VST plugins?

Audacity only hears system sounds. It is not capable of receiving microphone input. * In alsamixer, every option is unmuted and set to the maximum. * In PulseAudio Volume Control, the visual indicator clearly corresponds to the audio in my room, indicating that it correctly picks up my microphone. However, Audacity still only picks up sounds that come from other applications. It does not receive any microphone audio. I followed the [Arch Linux Wiki PulseAudio Troubleshooting Guide for Audacity](https://wiki.archlinux.org/title/PulseAudio/Troubleshooting#Audacity) . * pulseaudio-alsa is installed and configured to work for every application I've tried besides Audacity. * In Audacity, the only microphone selections are pulse and default. Both options only record computer sounds, not any microphone. * In PulseAudio Volume Control, I've tried all 12 permutations of settings from the Input Devices and Configurations tabs. None of these permutations allowed Audacity to pick up sound. * I tried running $ pasuspender -- audacity. This added more options for input devices besides pulse and default, but each option either recorded silence, halted the seeker, or crashed Audacity. I looked through the post on this forum titled [Microphone not showing up in Audacity](https://unix.stackexchange.com/questions/495407/microphone-not-showing-up-in-audacity) . * Similar to [menfon](https://unix.stackexchange.com/users/317995/menfon) , PulseAudio Volume Control clearly picked up on my microphone and demonstrated that my microphone works. Here is my output from $ amixer --card 0: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 3 [100%] [30.00dB] Front Right: 3 [100%] [30.00dB] Simple mixer control 'Mic Mute-LED Mode',0 Capabilities: enum Items: 'On' 'Off' 'Follow Capture' 'Follow Mute' Item0: 'Off' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 63 [100%] [30.00dB] [on] Front Right: Capture 63 [100%] [30.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Mute-LED Mode',0 Capabilities: enum Items: 'On' 'Off' 'Follow Master' Item0: 'Off' * I saw no parallels in my output that would allow me to use the same solution as [menfon](https://unix.stackexchange.com/users/317995/menfon) . I am using: * A Lenovo ThinkPad P14s Gen 1 * Arch Linux 5.12.12-arch1-1 * Audacity 2.4.1 * alsa-lib 1.2.5.1-3 * pulseaudio 14.2 * pulseaudio-alsa 1:1.2.5-2 Press me for any additional information you need. I greatly appreciate the help!

In windows it's very easy. How come in linux there is no solution? I have already tried using hda-jack-retask, but it was updated in 2014 last time and now its 2020 with Ubuntu 20.04 LTS. A lot has changed. So i tried it out (hda-jack-retask), which seemed to work but messed up with my Xfce4 desktop, and audacity was buggy/crashing/doing weird stuff. Never any of this happened before i use starting hdajackretask. Also i cant change the settings in PulseAudio control, becuase if i plug in my headset, the built in mic is automatically unplugged. There is no option to do this. Can someone help me?

I'm using Ubuntu 20.04 on an HP laptop and having trouble getting Audacity to record a track from line-in input. This may or may not be an issue with Audacity, it just acts like there's no line-in signal from my guitar amplifier and records the input from the built-in microphone instead. I tried following the advice here which suggested adding lines to /etc/pulse/default.pa based on the output of arecord -l. This had no effect than somehow disabling my built-in speakers, so I had to remove those lines and go back to the default. One more thing I tried was changing the "Profile" in the pavucontrol "Configuration" tab (see images linked at bottom of post) from "Analog Stereo Duplex" to "Analog Stereo Input", but that didn't work (and it wouldn't be great even if it did work, because I need to hear the click track through my headphones while playing the guitar and this option completely disables audio output from my computer.) I'm not great with ALSA and PulseAudio, but I think there might be an issue at the PulseAudio level because pavucontrol shows the microphone as "unplugged" (see images linked at bottom of post) even when my amplifier is plugged in. This isn't even an amplifier problem - it continues to say "unplugged" even when I connect my phone's output to my computer with a 3.5mm aux cable, and when I speak into the mic in my headphones. One more thing I tried was an "HDAJackRetask" tool which is supposed to fix faulty BIOS signals making the microphone appear unplugged (see images linked at bottom of post), but the interface doesn't really indicate to me what I should be changing to make the input signal go the right route. Any ideas? Edit: Somehow I've completely disabled my computer's internal microphone as well as line-in input - I was completely blocked from being able to talk on a video call yesterday.

The mic is visible in "PulseAudio volume control". It is not muted and volume is at 100%. I can see it picks sounds. It is also visible in arecord: $ arecord -l | tail -n 3 card 3: Microphone [Trust GXT 232 Microphone], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 $ arecord -L | tail -n 39 sysdefault:CARD=Microphone Trust GXT 232 Microphone, USB Audio Default Audio Device front:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio Front speakers surround21:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio IEC958 (S/PDIF) Digital Audio Output dmix:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio Direct sample mixing device dsnoop:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio Direct sample snooping device hw:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio Direct hardware device without any conversions plughw:CARD=Microphone,DEV=0 Trust GXT 232 Microphone, USB Audio Hardware device with all software conversions The mic is *not* available for selection in Audacity (I tried all options, launched Audacity *after* mic was connected). It works oob in Windows 10 in Audacity. I am using Audacity 2.1.2, PulseAudio 8 (Kubuntu 16.04). AlsaMixer: amixer: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 58 [91%] [-6.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [20.00dB] Front Right: 2 [67%] [20.00dB] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 0 [0%] [-64.00dB] [off] Front Right: Playback 0 [0%] [-64.00dB] [off] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 0 [0%] [-64.00dB] [off] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 0 [0%] [-64.00dB] [off] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'Line Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 46 [100%] [30.00dB] [on] Front Right: Capture 46 [100%] [30.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '4ch' '6ch' Item0: '2ch' Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Line' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Front Mic' 'Rear Mic' 'Line' Item0: 'Line' Simple mixer control 'Loopback Mixing',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Enabled' Simple mixer control 'Rear Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'Rear Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Not sure if the USB mic is there. I tried few other parameters where it seems to show up at the end. $ amixer controls: numid=35,iface=CARD,name='Front Headphone Jack' numid=31,iface=CARD,name='Front Mic Jack' numid=33,iface=CARD,name='Line Jack' numid=34,iface=CARD,name='Line Out Jack' numid=32,iface=CARD,name='Rear Mic Jack' numid=30,iface=MIXER,name='Master Playback Switch' numid=29,iface=MIXER,name='Master Playback Volume' numid=11,iface=MIXER,name='Headphone Playback Switch' numid=10,iface=MIXER,name='Headphone Playback Volume' numid=39,iface=MIXER,name='PCM Playback Volume' numid=26,iface=MIXER,name='Front Mic Boost Volume' numid=14,iface=MIXER,name='Front Mic Playback Switch' numid=13,iface=MIXER,name='Front Mic Playback Volume' numid=3,iface=MIXER,name='Front Playback Switch' numid=2,iface=MIXER,name='Front Playback Volume' numid=5,iface=MIXER,name='Surround Playback Switch' numid=4,iface=MIXER,name='Surround Playback Volume' numid=8,iface=MIXER,name='Center Playback Switch' numid=6,iface=MIXER,name='Center Playback Volume' numid=9,iface=MIXER,name='LFE Playback Switch' numid=7,iface=MIXER,name='LFE Playback Volume' numid=28,iface=MIXER,name='Line Boost Volume' numid=18,iface=MIXER,name='Line Playback Switch' numid=17,iface=MIXER,name='Line Playback Volume' numid=23,iface=MIXER,name='Capture Switch' numid=25,iface=MIXER,name='Capture Switch',index=1 numid=22,iface=MIXER,name='Capture Volume' numid=24,iface=MIXER,name='Capture Volume',index=1 numid=12,iface=MIXER,name='Loopback Mixing' numid=19,iface=MIXER,name='Auto-Mute Mode' numid=1,iface=MIXER,name='Channel Mode' numid=40,iface=MIXER,name='Digital Capture Volume' numid=20,iface=MIXER,name='Input Source' numid=21,iface=MIXER,name='Input Source',index=1 numid=27,iface=MIXER,name='Rear Mic Boost Volume' numid=16,iface=MIXER,name='Rear Mic Playback Switch' numid=15,iface=MIXER,name='Rear Mic Playback Volume' numid=37,iface=PCM,name='Capture Channel Map' numid=36,iface=PCM,name='Playback Channel Map' numid=38,iface=PCM,name='Capture Channel Map',device=2 $ amixer contents: numid=35,iface=CARD,name='Front Headphone Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=31,iface=CARD,name='Front Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=33,iface=CARD,name='Line Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=34,iface=CARD,name='Line Out Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=32,iface=CARD,name='Rear Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=30,iface=MIXER,name='Master Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=29,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=64,step=0 : values=58 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=11,iface=MIXER,name='Headphone Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=10,iface=MIXER,name='Headphone Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=64,step=0 : values=64,64 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=39,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=255,255 | dBscale-min=-51.00dB,step=0.20dB,mute=0 numid=26,iface=MIXER,name='Front Mic Boost Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0 : values=2,2 | dBscale-min=0.00dB,step=10.00dB,mute=0 numid=14,iface=MIXER,name='Front Mic Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=13,iface=MIXER,name='Front Mic Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0 : values=0,0 | dBscale-min=-34.50dB,step=1.50dB,mute=0 numid=3,iface=MIXER,name='Front Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=2,iface=MIXER,name='Front Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=64,step=0 : values=64,64 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=5,iface=MIXER,name='Surround Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=off,off numid=4,iface=MIXER,name='Surround Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=64,step=0 : values=0,0 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=8,iface=MIXER,name='Center Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=6,iface=MIXER,name='Center Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=64,step=0 : values=0 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=9,iface=MIXER,name='LFE Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=7,iface=MIXER,name='LFE Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=64,step=0 : values=0 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=28,iface=MIXER,name='Line Boost Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0 : values=0,0 | dBscale-min=0.00dB,step=10.00dB,mute=0 numid=18,iface=MIXER,name='Line Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=17,iface=MIXER,name='Line Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0 : values=0,0 | dBscale-min=-34.50dB,step=1.50dB,mute=0 numid=23,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=25,iface=MIXER,name='Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=2 : values=off,off numid=22,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=46,step=0 : values=46,46 | dBscale-min=-16.00dB,step=1.00dB,mute=0 numid=24,iface=MIXER,name='Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=46,step=0 : values=0,0 | dBscale-min=-16.00dB,step=1.00dB,mute=0 numid=12,iface=MIXER,name='Loopback Mixing' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Disabled' ; Item #1 'Enabled' : values=1 numid=19,iface=MIXER,name='Auto-Mute Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Disabled' ; Item #1 'Enabled' : values=0 numid=1,iface=MIXER,name='Channel Mode' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 '2ch' ; Item #1 '4ch' ; Item #2 '6ch' : values=0 numid=40,iface=MIXER,name='Digital Capture Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=120,step=0 : values=60,60 | dBscale-min=-30.00dB,step=0.50dB,mute=0 numid=20,iface=MIXER,name='Input Source' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Front Mic' ; Item #1 'Rear Mic' ; Item #2 'Line' : values=2 numid=21,iface=MIXER,name='Input Source',index=1 ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Front Mic' ; Item #1 'Rear Mic' ; Item #2 'Line' : values=2 numid=27,iface=MIXER,name='Rear Mic Boost Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0 : values=0,0 | dBscale-min=0.00dB,step=10.00dB,mute=0 numid=16,iface=MIXER,name='Rear Mic Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=15,iface=MIXER,name='Rear Mic Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0 : values=0,0 | dBscale-min=-34.50dB,step=1.50dB,mute=0 numid=37,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=36,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=38,iface=PCM,name='Capture Channel Map',device=2 ; type=INTEGER,access=r----R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR

I have a very noisy heater in my apartment producing a constant whining sound between 6230 and 6280 Hz. The frequency varies but seems to fall within this range. Other apartments in my building have this same problem and the landlord doesn't seem too interested in fixing it. The noise is very annoying so I thought I could use some noise cancellation software to effectively cancel the noise. Using Audacity, I took a recording of the noise, and followed this tutorial http://wiki.audacityteam.org/wiki/Noise_Removal and was able to remove the noise from the audio file. Now, my question is, how can I do this in real time? I'd like to produce some "anti-noise" to cancel out the whining sound. Searching around for this, I found lots of people trying to do real time cancellation of background noise for Skype or other applications where the cancellation is being applied to an audio stream (or file) which is then being listened to elsewhere. I can't find any examples of this being done to cancel the noise in the location where it is being heard. Is this even possible? Could this be done with a tool such as Ardour?

**UPDATE:** Today's upgrade to alsa-lib-1.1.7-2 fixed the issue. (2018-11-23) ---------- Arch Linux, up-to-date. As so often, a -Syu upgrade broke things again. This time it's audacity. Usually audacity, and **any** other playback/record app would show up in pauvcontrol - Not audacity anymore. I could use pauvcontrol to choose sink and source from which I want to record or playback audio using audacity - not anymore. In audacity's preferences it only says ALSA in the top dropdown, im not sure if that is supposed to be like that or if it said PulseAudio before. There aren't even all my devices in the playback tab. Same goes for the recording device selection, but the loopback device I need to record from is there, so that works. But now I wanted to playback sound from audacity too and it just doesn't work anymore. It doesn't list my USB playback device and when I set it to sysdefault I just get an error. $ pulseaudio -v pulseaudio 12.2 $ audacity --version lilv_world_add_plugin(): warning: Duplicate plugin lilv_world_add_plugin(): warning: ... found in file:///usr/lib/lv2/eg-params.lv2/ lilv_world_add_plugin(): warning: ... and file:///usr/lib64/lv2/eg-params.lv2/ (ignored) lilv_world_add_plugin(): warning: Duplicate plugin lilv_world_add_plugin(): warning: ... found in file:///usr/lib/lv2/eg-midigate.lv2/ lilv_world_add_plugin(): warning: ... and file:///usr/lib64/lv2/eg-midigate.lv2/ (ignored) lilv_world_add_plugin(): warning: Duplicate plugin lilv_world_add_plugin(): warning: ... found in file:///usr/lib/lv2/eg-fifths.lv2/ lilv_world_add_plugin(): warning: ... and file:///usr/lib64/lv2/eg-fifths.lv2/ (ignored) lilv_world_add_plugin(): warning: Duplicate plugin lilv_world_add_plugin(): warning: ... found in file:///usr/lib/lv2/eg-metro.lv2/ lilv_world_add_plugin(): warning: ... and file:///usr/lib64/lv2/eg-metro.lv2/ (ignored) lilv_world_add_plugin(): warning: Duplicate plugin lilv_world_add_plugin(): warning: ... found in file:///usr/lib/lv2/eg-amp.lv2/ lilv_world_add_plugin(): warning: ... and file:///usr/lib64/lv2/eg-amp.lv2/ (ignored) lilv_world_add_plugin(): warning: Duplicate plugin lilv_world_add_plugin(): warning: ... found in file:///usr/lib/lv2/eg-sampler.lv2/ lilv_world_add_plugin(): warning: ... and file:///usr/lib64/lv2/eg-sampler.lv2/ (ignored) lilv_world_add_plugin(): warning: Duplicate plugin lilv_world_add_plugin(): warning: ... found in file:///usr/lib/lv2/eg-scope.lv2/ lilv_world_add_plugin(): warning: ... and file:///usr/lib64/lv2/eg-scope.lv2/ (ignored) lilv_world_add_plugin(): warning: Duplicate plugin lilv_world_add_plugin(): warning: ... found in file:///usr/lib/lv2/eg-scope.lv2/ lilv_world_add_plugin(): warning: ... and file:///usr/lib64/lv2/eg-scope.lv2/ (ignored) lilv_world_add_plugin(): warning: Reloading plugin lilv_world_add_plugin(): warning: Reloading plugin lilv_world_add_plugin(): warning: Reloading plugin lilv_world_add_plugin(): warning: Reloading plugin lilv_world_add_plugin(): warning: Reloading plugin lilv_world_add_plugin(): warning: Reloading plugin lilv_world_add_plugin(): warning: Reloading plugin lilv_world_add_plugin(): warning: Reloading plugin Not sure if those warnings mean anything, it won't print the version. The package is audacity-2.3.0-1-x86_64.pkg.tar.xz Full log of the console when starting audacity: ALSA lib pcm_dmix.c:1099:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.iec958.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM iec958 ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.iec958.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM spdif ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.iec958.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM spdif ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.hdmi.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM hdmi ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.hdmi.0:CARD=0,AES0=4,AES1=130,AES2=0,AES3=2' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM hdmi ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.modem.0:CARD=0' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline:CARD=0,DEV=0 ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.modem.0:CARD=0' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline:CARD=0,DEV=0 ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.modem.0:CARD=0' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM phoneline ALSA lib confmisc.c:1281:(snd_func_refer) Unable to find definition 'cards.Loopback.pcm.modem.0:CARD=0' ALSA lib conf.c:4555:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5034:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM phoneline connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed ALSA lib pcm_oss.c:377:(_snd_pcm_oss_open) Unknown field port ALSA lib pcm_oss.c:377:(_snd_pcm_oss_open) Unknown field port Expression 'alsa_snd_pcm_hw_params_set_period_size_near( pcm, hwParams, &alsaPeriodFrames, &dir )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 924 ALSA lib pcm_usb_stream.c:486:(_snd_pcm_usb_stream_open) Invalid type for card ALSA lib pcm_usb_stream.c:486:(_snd_pcm_usb_stream_open) Invalid type for card ALSA lib pcm_dmix.c:1099:(snd_pcm_dmix_open) unable to open slave Expression 'alsa_snd_pcm_hw_params_set_period_size_near( pcm, hwParams, &alsaPeriodFrames, &dir )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 924 connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed Even though I'm not sure if those errors are because of ALSA and maybe completely unrelated to the PulseAudio issue.

I was trying to get the Gentoo version from the Audacity website, but as far as I can tell, Audacity is not ready for Gentoo Linux yet. Anyone know if the Ubuntu version will work on Alpine? Or one of the others?

ever since I upgraded Debian stable to Debian sid I cannot start audacity. It gives me this error when I start from command line: audacity: relocation error: /usr/lib/x86_64-linux-gnu/libwx_gtk2u_adv-3.0.so.0: symbol _ZThn696_N15wxComboCtrlBase9WriteTextERK8wxString version WXU_3.0 not defined in file libwx_gtk2u_core-3.0.so.0 with link time reference I still cannot find out how to fix this problem.

Is it possible to convert a mp4 file to mp3 or flac, clearing background noises in the process? Or is it possible to run audacity totally trough shell, No GUI?

This is how I have defined my soundcard's channels in .asoundrc: pcm.!default { type plug slave { pcm "hw:1,0" } } ctl.!default { type hw card 1 } pcm_slave.usb_1 { pcm "hw:1,0" rate 44100 channels 8 buffer_size 4096 period_size 1024 } pcm.outch1 { type dshare ipc_key 1111 slave usb_1 bindings [ 0 ] hint.description "USB output/playback channel 1 (from output port 1)" } pcm.inch1 { type dsnoop ipc_key 1111 slave usb_1 bindings [ 0 ] hint.description "USB input/capture channel 1 (from input port 1)" } While the above **works fine with aplay and arecord** functions/commands in the terminal, trying to play a 8k audio sample with Audacity throws the following error : When I check the Audio device info option in Help menu (Help-->Audio device option), I see the following: Device ID: 2 Device name: USB Audio (hw:1,0) Host name: ALSA Recording channels: 8 Playback channels: 8 Low Recording Latency: 0.008707 Low Playback Latency: 0.008707 High Recording Latency: 0.034830 High Playback Latency: 0.034830 Supported Rates: 44100 48000 88200 96000 ============================== Device ID: 10 Device name: outch1 Host name: ALSA Recording channels: 0 Playback channels: 1 Low Recording Latency: -1.000000 Low Playback Latency: 0.023220 High Recording Latency: -1.000000 High Playback Latency: 0.023220 Supported Rates: 44100 ============================== Device ID: 11 Device name: inch1 Host name: ALSA Recording channels: 1 Playback channels: 0 Low Recording Latency: 0.023220 Low Playback Latency: -1.000000 High Recording Latency: 0.023220 High Playback Latency: -1.000000 Supported Rates: ============================== As we can see, 8000 or 16000 is not a Supported Rate from the audio device info. So, it's quite clear why the error is thrown. So, how and where can I add/set (modify my .asoundrc maybe?) these sample rates so that any application (including Audacity of course) can support playback and recording of audio files at any desired sample rate.

I'm trying to use a dark Gtk+ theme, and everything goes fine with native Gtk+ apps, and well, only with Gtk+ apps. I mean, wxWidgets and Mozilla's apps looks terrible, if not counting unreadability (see screenshot below). Rendering Gtk's buttons and fonts in Qt apps looks bad too. I've learned Mozilla's apps could improve readability (text on light background become readable **but** text on dark background becomes hard to read) by adding this to gtk-2.0/gtkrc: style "moz" { text[NORMAL] = "#000000" base[NORMAL] = "#FFFFFF" } class "GtkInvisible" style "moz" apwidget "GtkInvisible" style "moz" So I came out with this solution: start these apps with a 'fallback' light Gtk+ 2 theme (let's say Adwaita). But launching with something like GTK2_RC_FILES=/usr/share/themes/Adwaita/gtk-2.0/gtkrc could change gtk's PNGs (like buttons and checkboxes), **not** the color scheme: I noted that doing this, it returns some warnings, e.g.: $ GTK2_RC_FILES=/usr/share/themes/Adwaita/gtk-2.0/gtkrc audacity (Audacity:12322): Gtk-WARNING **: gtk_disable_setlocale() must be called before gtk_init() $ GTK2_RC_FILES=/usr/share/themes/Adwaita/gtk-2.0/gtkrc icedove (process:13204): GLib-CRITICAL **: g_slice_set_config: assertion 'sys_page_size == 0' failed With Qt(4) apps, I tried to use a native good-looking theme like Breeze, but couldn't find a way to choose it after installing (I had no luck using qtconfig-qt4). Here are some of my software versions (I'm using Debian testing): - Gtk+ 2: 2.24.28 - Gtk+ 3: 3.16.6 - Qt4: 4.8.7 - wxWidgets: 3.0.2 (Audacity's version is 2.0.6) - Iceweasel (Firefox): 38.1.0 - Icedove (Thunderbird): 31.7.0 Does anyone know how to solve this problem? I would be really glad to see any answer, even if it's specific to only one application.

I am trying to write a script file and invoke it using XFCE global shortcut. The script should continue the Audacity application if it is in the stopped state and stop it (pause recording) if it is in the run state. This will be useful when I have defined the same keyboard shortcut for VLC's playback/pause. This way, I can do playback/pause (in VLC) and record/pause (in Audacity) at the same time using the same keyboard shortcut. Getting some ideas from [this post](https://askubuntu.com/questions/855893/is-it-possible-to-control-the-recording-if-audacity-is-running-in-the-background) I have written the following script and added it to XFCE's custom keyboard settings. But it does not work. #!/bin/bash if pgrep -f "audacity" ; then pkill -stop audacity && notify-send "Recording stopped" else pkill -cont audacity