I have *Ubuntu 19.04*, and I am facing issues with sounds. When playing audio, I get audio through only one of the speakers (the left one, that is). Also, the volume is pretty low. I run the same on windows 10 (through dual boot), and both speakers run perfectly fine with proper amplification. What should I do?

I use pulseaudio on my system. I have installed the tts applications espeak and flite. Both of which work perfectly when called from the commandline, but fail to work when called via spd-say. I need them to work via spd-say, because I want to configure a TTS engine for Okular. Here is the output for sudo systemctl status speech-dispatcherd.service

● speech-dispatcherd.service - Speech-Dispatcher an high-level device independent layer for speech synthesis.
   Loaded: loaded (/usr/lib/systemd/system/speech-dispatcherd.service; disabled; vendor preset: disabled)
   Active: failed (Result: exit-code) since Fri 2019-01-18 00:30:18 IST; 2min 9s ago
  Process: 1067 ExecStart=/usr/bin/speech-dispatcher -d (code=exited, status=1/FAILURE)

Jan 18 00:30:18 archlinux-hostname systemd: Starting Speech-Dispatcher an high-level device independent layer for speech synthesis....
Jan 18 00:30:18 archlinux-hostname speech-dispatcher: [Fri Jan 18 00:30:18 2019 : 363344] speechd: Speech Dispatcher 0.8.8 starting
Jan 18 00:30:18 archlinux-hostname systemd: speech-dispatcherd.service: Control process exited, code=exited, status=1/FAILURE
Jan 18 00:30:18 archlinux-hostname systemd: speech-dispatcherd.service: Failed with result 'exit-code'.
Jan 18 00:30:18 archlinux-hostname systemd: Failed to start Speech-Dispatcher an high-level device independent layer for speech synthesis..

Here is the output for journalctl -xe

➜ journalctl -xe

Jan 18 00:30:18 archlinux-hostname sudo:       codingcoffee : TTY=pts/1 ; PWD=/home/codingcoffee ; USER=root ; COMMAND=/usr/bin/systemctl restart speech-dispatcherd.service
Jan 18 00:30:18 archlinux-hostname sudo: pam_unix(sudo:session): session opened for user root by codingcoffee(uid=0)
Jan 18 00:30:18 archlinux-hostname systemd: Starting Speech-Dispatcher an high-level device independent layer for speech synthesis....
-- Subject: A start job for unit speech-dispatcherd.service has begun execution
-- Defined-By: systemd
-- Support: https://lists.freedesktop.org/mailman/listinfo/systemd-devel 
-- 
-- A start job for unit speech-dispatcherd.service has begun execution.
-- 
-- The job identifier is 4254.
Jan 18 00:30:18 archlinux-hostname speech-dispatcher: [Fri Jan 18 00:30:18 2019 : 363344] speechd: Speech Dispatcher 0.8.8 starting
Jan 18 00:30:18 archlinux-hostname systemd: speech-dispatcherd.service: Control process exited, code=exited, status=1/FAILURE
-- Subject: Unit process exited
-- Defined-By: systemd
-- Support: https://lists.freedesktop.org/mailman/listinfo/systemd-devel 
-- 
-- An ExecStart= process belonging to unit speech-dispatcherd.service has exited.
-- 
-- The process' exit code is 'exited' and its exit status is 1.
Jan 18 00:30:18 archlinux-hostname systemd: speech-dispatcherd.service: Failed with result 'exit-code'.
-- Subject: Unit failed
-- Defined-By: systemd
-- Support: https://lists.freedesktop.org/mailman/listinfo/systemd-devel 
-- 
-- The unit speech-dispatcherd.service has entered the 'failed' state with result 'exit-code'.
Jan 18 00:30:18 archlinux-hostname systemd: Failed to start Speech-Dispatcher an high-level device independent layer for speech synthesis..
-- Subject: A start job for unit speech-dispatcherd.service has failed
-- Defined-By: systemd
-- Support: https://lists.freedesktop.org/mailman/listinfo/systemd-devel 
-- 
-- A start job for unit speech-dispatcherd.service has finished with a failure.
-- 
-- The job identifier is 4254 and the job result is failed.
Jan 18 00:30:18 archlinux-hostname audit: SERVICE_START pid=1 uid=0 auid=4294967295 ses=4294967295 msg='unit=speech-dispatcherd comm="systemd" exe="/usr/lib/systemd/systemd" hostname=? addr=? terminal=? res=fail>
Jan 18 00:30:18 archlinux-hostname kernel: audit: type=1130 audit(1547751618.383:189): pid=1 uid=0 auid=4294967295 ses=4294967295 msg='unit=speech-dispatcherd comm="systemd" exe="/usr/lib/systemd/systemd" hostname=>
Jan 18 00:30:18 archlinux-hostname sudo: pam_unix(sudo:session): session closed for user root
Jan 18 00:30:19 archlinux-hostname sudo:       codingcoffee : TTY=pts/1 ; PWD=/home/codingcoffee ; USER=root ; COMMAND=/usr/bin/systemctl status speech-dispatcherd.service
Jan 18 00:30:19 archlinux-hostname sudo: pam_unix(sudo:session): session opened for user root by codingcoffee(uid=0)
Jan 18 00:30:19 archlinux-hostname sudo: pam_unix(sudo:session): session closed for user root

Here is the output to my speech-dispatcher config file ~/.config/speech-dispatcher/speechd.conf

LogLevel  3
LogDir  "default"
DefaultRate   0
DefaultPitch   0  
DefaultVolume 100
DefaultLanguage   en
AudioOutputMethod   pulse
DefaultModule   espeak
Include "clients/*.conf"

I'm running Debian 10 on my laptop. I tried to install a driver for the realtek alc665 sound card. I failed and as the result I don't have sound on my system.I can play songs but I can't hear anything though volume control is available and HIGH. When I run

audio mixer comes up and in the configuration tab I see:

cards available for configuration

In the output devices tab I see

output

I opened up a terminal and this is the output from

-v

: (Only audio devices in this list) 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) Subsystem: Dell 5 Series/3400 Series Chipset High Definition Audio Flags: fast devsel, IRQ 11 Memory at f0a00000 (64-bit, non-prefetchable) [size=16K] Capabilities: Kernel modules: snd_hda_intel 02:00.1 Audio device: NVIDIA Corporation GF108 High Definition Audio Controller (rev a1) Subsystem: Dell GF108 High Definition Audio Controller Flags: bus master, fast devsel, latency 0, IRQ 7 Memory at cdefc000 (32-bit, non-prefetchable) [size=16K] Capabilities: Kernel modules: snd_hda_intel The output form

-c multimedia

might also be usefull: *-multimedia UNCLAIMED description: Audio device product: GF108 High Definition Audio Controller vendor: NVIDIA Corporation physical id: 0.1 bus info: pci@0000:02:00.1 version: a1 width: 32 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: latency=0 resources: memory:cdefc000-cdefffff *-usb:1 description: Video product: Laptop_Integrated_Webcam_2HDM vendor: CN07CN2C786640CA03F0A00 physical id: 4 bus info: usb@1:1.4 version: 9.01 capabilities: usb-2.00 configuration: driver=uvcvideo maxpower=250mA speed=480Mbit/s *-multimedia UNCLAIMED description: Audio device product: 5 Series/3400 Series Chipset High Definition Audio vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 06 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress cap_list configuration: latency=0 resources: memory:f0a00000-f0a03fff When I ran alsa –version , I got this: ALSA lib confmisc.c:767:(parse_card) cannot find card '0' ALSA lib conf.c:4568:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:4568:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name ALSA lib conf.c:4568:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory ALSA lib conf.c:5047:(snd_config_expand) Evaluate error: No such file or directory ALSA lib pcm.c:2565:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:828: audio open error: No such file or directory and finally this is the output from aplay -l: aplay: device_list:272: no soundcards found... It seems that drivers are currently available but not for pulseaudio. So how could I fix this?

I run Mopidy as a service for my mpd because of the nice Spotify integration. I've followed the documentation's instructions to use it with pulseaudio by setting

load-module module-native-protocol-tcp auth-ip-acl=127.0.0.1`

in my /etc/pulse/default.pa and

[audio]
output = pulsesink server=127.0.0.1

in my /etc/mopidy/modipy.conf.
All of this was working perfectly, however recently I've started using Pipewire as a replacement for PulseAudio on Arch Linux by following this and installing pipewire,pipewire-alsa, pipewire-pulse and pipewire-media-session. But now I've noticed that whenever I try to play any music nothing happens. I checked the journalctl for Mopidy and found this

ERROR ... [536:MainThread] mopidy.audio.gst
... mopidy:   GStreamer error: Failed to connect: Connection refused
... mopidy: WARNING ... [536:Audio-2] mopidy.audio.actor
... mopidy:   Setting GStreamer state to GST_STATE_PLAYING failed
... mopidy: WARNING ... [536:Core-11] mopidy.core.tracklist
... mopidy:   Track is not playable: local:track:...

I searched around and found the gst-plugin-pipewire package, but installing it didn't help at all. Apart from Mopidy, everything else audio-wise is working perfectly. If I could get some help with this I'd be thankful.

I'm using pulseaudio with a 6-channel USB sound device (which reports itself as a "CM106 Like Sound Device"). I added the following to ~/.config/pulse/daemon.conf: enable-lfe-remixing = yes lfe-crossover-freq = 50 If I open pavucontrol and, under the configuration tab, select "Analog Surround 5.1 Output," everything works fine as best I can tell, except for the fact that I don't have 5 speakers. (I can test by moving the speakers around to different output ports and playing surround sound test files.) Unfortunately, if I select either "Analog Surround 4.1 Output" or "Analog Surround 2.1 Output", I get no sound out of my subwoofer. Is there any way to make my subwoofer work with fewer than 6 channels?

I've installed Kali 2023.1. Although I tried all sound driver, sound not working. But when I plugged any device like headphone It's working. I also tried Kali other version and Ubuntu latest version but It didn't work. My laptop has dual boot Windows 11 and Kali. GPU: RTX 4080 Mobile CPU: i9 13900hx I have tried these steps: https://www.kali.org/docs/troubleshooting/no-sound/ https://stevemats.medium.com/fix-sound-issue-in-kali-linux-2021-4-babbed3cce0 https://ourcodeworld.com/articles/read/406/how-to-enable-the-sound-on-kali-linux https://unix.stackexchange.com/a/643308 https://askubuntu.com/a/1400275 https://askubuntu.com/a/1277167 lspci | grep -i audio 00:1f.3 Audio device: Intel Corporation Device 7a50 (rev 11) 01:00.1 Audio device: NVIDIA Corporation Device 22bc (rev a1)

I have a digital piano connected to a Focusrite Scarlett 2i2 USB audio interface. That interface and a Logitech USB webcam are both connected to my computer. I want to merge the sounds from the Scarlett and the webcam microphone into a null-sink and use its monitor as "microphone" in Skype for remote Piano lessons. The teacher can then hear me speak and also the music directly from the Scarlett (and not recorded via air). So far I do this: pactl load-module module-null-sink sink_name=piano_lesson pactl load-module module-loopback sink=piano_lesson pactl load-module module-loopback sink=piano_lesson So now I have to loopbacks going into the null-sink. I then open pavucontrol GUI and on the recording tab I assign the Scarlett to the first loopback and the webcam mic to the second loopback. This works. However, this is tedious and I have begun writing a script to automate the process. However, I struggle to find the command for assigning the devices to those loopbacks. I'm not sure if pactl move-sink-input|move-source-output are the right commands for this nor have I fully understood their purpose even through reading documentation and other Stack questions/answers. The commands provide auto-completion but the suggestions for the parameters are confusing or expected parameters (e.g. the microphone) do not show up. **Q: How would I connect the Scarlett/microphone to the loopback devices via command line?** My goal: - connect source #4 (alsa_input.usb-Focusrite_Scarlett_2i2_[...].analog-stereo) to sink-input 1 - connect source #5 (alsa_input.usb-046d_09a1_C5BDCE20-02.analog-mono [=webcam]) to sink-input 2 Here are some outputs: $ pactl list sinks short 0 alsa_output.pci-0000_00_1b.0.analog-stereo module-alsa-card.c s16le 2ch 44100Hz SUSPENDED 1 alsa_output.usb-Focusrite_Scarlett_2i2_USB_Y8GCE759613412-00.analog-stereo module-alsa-card.c s32le 2ch 44100Hz SUSPENDED 2 piano_lesson module-null-sink.c s16le 2ch 44100Hz RUNNING $ pactl list sink-inputs short 0 0 1 protocol-native.c float32le 2ch 44100Hz 1 2 - module-loopback.c s16le 2ch 44073Hz 2 2 - module-loopback.c s16le 2ch 44032Hz $ pactl list sources short 0 alsa_output.pci-0000_00_1b.0.analog-stereo.monitor module-alsa-card.c s16le 2ch 44100Hz SUSPENDED 1 alsa_input.pci-0000_00_1b.0.analog-stereo module-alsa-card.c s16le 2ch 44100Hz SUSPENDED 3 alsa_output.usb-Focusrite_Scarlett_2i2_USB_Y8GCE759613412-00.analog-stereo.monitor module-alsa-card.c s32le 2ch 44100Hz SUSPENDED 4 alsa_input.usb-Focusrite_Scarlett_2i2_USB_Y8GCE759613412-00.analog-stereo module-alsa-card.c s32le 2ch 44100Hz RUNNING 5 alsa_input.usb-046d_09a1_C5BDCE20-02.analog-mono module-alsa-card.c s16le 1ch 16000Hz RUNNING 6 piano_lesson.monitor module-null-sink.c s16le 2ch 44100Hz IDLE $ pactl list source-outputs short 7 4 - module-loopback.c s16le 2ch 44100Hz 9 5 - module-loopback.c s16le 2ch 44100Hz **Note:** I'm also open for non-PulseAudio solutions and it doesn't matter if it is a script or a GUI based tool in which I could load such a configuration, e.g. from a profile.

I'm running XUbuntu 16.04. Previously I had speakers connected to analog line-out, and sound would automatically switch over to headphones when they were plugged in. This was all happening on one output device (motherboard audio). I've now got a new monitor with built-in speakers, but no analog connection, so I'm getting audio over DisplayPort from my Radeon R9 270. I'd like pulseaudio to switch existing streams and new streams to the headphone port on the motherboard audio when I plug in the headphones, and back to the DP audio port on the GPU device when I unplug them again. I've tried using pactl load-module module-switch-on-connect, but it doesn't seem to have any effect, presumably because plugging in the headphones doesn't create a new sink, only a new port. I know from reading other questions that I can do this manually with pactl/pacmd invocation and I'll do that if I have to, but I'd much prefer to have this happen automatically. I don't mind if I have to do some scripting to do it, but is there a hook I can intercept to get a call when the headphones are connected or disconnected?

Currently running Pop!_OS 22.04 LTS X64 I'm extremely new to linux as a whole so I'm sorry if this is a stupid question. I'm attempting to set up virtual audio cables for recording and I've managed to find that using pulseaudio I can have them, but it will not persist through a reboot. I was told to make a file called default.pa in ~/.config/pulse directory and what I have in there at the moment is

.include /etc/pulse/default.pa
.ifexists module-virtual-sink.so
load-module module-virtual-sink sink_name=VAC_1
load-module module-virtual-sink sink_name=VAC_2
load-module module-virtual-sink sink_name=VAC_3

but it was not working after a reboot and even attempting to format it the same as the command I run in Terminal to make them. The command I am running to make the audio channels normally is

pactl load-module module-virtual-sink sink_name=VAC_1

I'm sure I am doing something wrong but, as I am inexperienced, I am lost.

I am reading about pulseaudio, how it works and how I can configure it. I am encountering two keywords a lot: ***SINK***, ***SOURCE.*** At first I thought ***SINK*** meant ***OUTPUT*** and ***SOURCE*** meant ***INPUT***, but it seems that this is not the case. Could someone explain what ***SINK*** and ***SOURCE*** mean in simple English?

OS is Debian 12. Every now and then, in pavucontrol I'll see: "Establishing connection to PulseAudio. Please wait..." So it looks like pulseaudio crashed. I restarted pulseaudio with pulseaudio -k and everything looks good in pavucontrol, but all my audio programs are now missing from the playback tab. I'm guessing they "lost the connection" to pulseaudio when it crashed. How do I restore these connections without having to restart each program? I have a few dozen audio programs that were connected to pulseaudio before the crash, and I really don't want to have to restart each one, as restoring the full state/workflow of each program would take quite a bit of time.

I have a hp laptop 15s fq as my main pc and from the moment I started using it, I had many problems. Im asking here because sadly I had no luck on zorin forums and I was wondering if there is something I can do Firstly, I have a intel i5 and an intel irisxe gpu and from the moment I installed zorin on it, I was immediately met with tiny scaling on 1920x1080 and it wasnt usable. So my only solution was fractional scaling to 125% witch worked fine until I was met with screen tearing on xorg. Wayland was a temporary fix as x11 apps on wayland where blurry A guy at hp community told me to edit graphics configuration like this

Section "Device"    
    
    Identifier  "Intel Graphics"    
    
    Driver      "intel"    
    
    Option      "TearFree" "true"    
    
    Option      "AccelMethod" "sna"    
    
    Option      "DRI" "3"    
    
    Option      "TripleBuffer" "true" EndSection

But this only made it worse because it froze x11 and I had to force reboot Secondly, I started having severe slowdown and distorted audio while I was charging or doing things like playing some lightweight discord games which I didn't expect to do this... Sometimes audio gets distorted for now reason at all. As a dev having such slowdown while using many windows open it is bad...Its like my keyboard has a delay.. Other times it just doesn't have problems at all *when its mostly not charging. I have to note that I have a nvme ssd and ddr16gb ram so I don't really understand the reason it does that Can anyone help?

**Issue** I cannot capture audio using an external microphone on my Linux Mint 20 Ulyana system. In Sound Settings under the Input tab, when I select the external microphone from the Device list, the Input level shows zero/nil, irrespective of the Volume slider position. Sound Settings detailing the above issue." class="img-fluid rounded" style="max-width: 100%; height: auto; margin: 10px 0;" loading="lazy"> However, I can still use the internal microphone without any issue. **Things I have tried so far** 1. I have tried three different external microphones, including a bluetooth device. I have eliminated the possibility of an issue with Microphone jack and the microphone pin(s). 2. Checked whether the device is muted in alsamixer. It looks like the device is unmuted, as I did not see an MM under the device. Here's my system information: https://termbin.com/31ml

I am trying to redirect the sound produced by laptop to a bluetooth speaker. I know how to redirect the sound of sink inputs to different sinks. However, when I am playing music, sound is coming from my laptop but when I use the command $ pacmd list-sink-inputs I get this in response 0 sink input(s) available. I have no idea why I am getting this response. I have tested with multiple applications but I continue to receive this. Does anyone have ideas where I can look into this to investigate this further?

I successfully configured PulseAudio server and client to send audio over network. It uses direct connection: http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Network/#index1h2 I'd like to have a possibility to switch between client and server sound card i.e. temporarily disable network stream and go back to internal sound device. Using module-tunnel-sink I could simply move sink-input to desired device but is not an option since it doesn't work well with Flash: >they lead me to believe that Flash is somehow sending the sound to PulseAudio in such a way that it creates a lot of network traffic (think lots of tiny packets, not bandwidth); this overwhelms the network "tunnel" PulseAudio With direct connection I have to restart the application every time I want to switch the output. Any idea how can I solve this?

I have Centos 6.3 running on this pc. The built in sound card is detected normally and is a 7.1 card with all speakers listed. It includes a optical out as well as HDMI for the built in video. It looks like pulse audio is automagically misconfiguring the optical out, as it is listed as stereo digital out. The analog out is listed as analog stereo - analog surround 5.1. There are 4 or so different variations listed in the pulseaudio sound applet. If I select the test sound it has left and right front. I get audio through the digital connection but is only those 2 channels. opening alsamixer it appears that the surround is activated and detected. It defaults to pulseaudio for both input and output. If I select the hardware card instead, it appears they are turned on properly. I need a way to specify that digital audio is more than 2 channels, or create a specific profile for sound through the iec958 link with 8 channels.

Debian 12. If I try and play a video through my bluetooth headset, the video freezes on launch. Tried both mpv and vlc. If I use pavucontrol and select the motherboard sound card, the video plays without any problems and the audio is just fine. If I toggle back and forth between the two, the video will stop and start. I've tried restarting bluetooth and pulseaudio, but no luck. My headset has worked fine for years, this is the first time this has happened. How do I fix this without rebooting?

The sound configuration has spontaneously stopped working on my Linux laptop. I do not know the cause, or how to fix it. ### Symptoms: In the PulseAudio control panel, I can set Built-in Audio to - Analog Stereo Input - Pro Audio - Various Digital Surround... options that all say "(unavailable) (unplgged)" - Off If I use Analog Stereo Input, my laptop microphone works but my laptop cannot play sound. If I use Pro Audio, my laptop can play sound (seemingly at a lower volume than it used to be capable of), but my microphone does not work. I am not sure when this issue appeared, but it most likely entered with a routine apt-get update some time in May of 2025. To my recollection, I have not manually installed any new packages during this time, but it is possible that an unrelated package update has changed the audio configuration. Prior to this, the Built-in Audio device could play audio through the laptop speakers and record audio through the laptop microphone at the same time. This worked out of the box with, for example, video calls, from the day I installed Debian up until a few weeks ago. I did not need to manually adjust any settings for video calls to work. ### System: - Debian (12.11) Linux - Lenovo P51 laptop - Kernel version 6.1.0-34-amd64 - All standard update applied via apt as of 01 June 2025. - Single boot device, no other operating systems. - The system appears to be using PulseAudio, and I have not manually altered the audio configuration since the original (stock) installation of Debian 12 approximately 6 months ago. I have never been very skilled in understanding, diagnosing, debugging, or fixing audio issues in my 20 years of using Linux, so I thank you for your patience and advice in advance. I am not sure what steps to take next to try to fix this.

I used Realtek Audio manager in case of Windows and I was able to use my the microphone that is integrated with my head-phones. I have installed PulseAudio Volume Control. But it doesn't show up in the input devices section. Edit : I want the system to detect it as a headset rather than a headphone. Here's the screenshot of PulseAudio Here's the input device list. Only microphone is present in the drop down list.

EDIT: Ok so it appears to be working... kind of... sometimes the cancellation kicks in, other times it doesn't within the same call. This is not solved would still love some input. *** 2nd EDIT: I've posted in the Pulseaudio IRC so hopefully get some advice on this setup. *** I have pulseaudio loading the module-echo-cancel and it doesn't seem to be working properly, or it is and I'm not doing something right... I honestly have no way to confirm which I suppose is my first issue. My architecture is fairly simple, I have a windows 10 machine that I'm rdp'ing into from Linux. Default source: - ATR (with echo cancel at HDMI) Default Sink: - HDMI (with echo cancel at ATR) I shouldn't think that remote access would affect the cancellation as it's still being processed locally from the HDMI speakers directly into the microphone, both of which are local devices. Here is how I am loading module-echo-cancel in ~/.config/pulse/default.pa and I can only assume it is loading given that I can see the devices (as listed above) in pasystray. .ifexists module-echo-cancel.so load-module module-echo-cancel source_master=alsa_input.usb-Conference_USB_microphone_ATR4697-USB-00.mono-fallback sink_master=alsa_output.pci-0000_00_03.0.hdmi-stereo source_name=echocancel sink_name=echocancel1 set-default-source echocancel set-default-sink echocancel1 .endif Are there any extra controls I need to consider or should this work OOTB? I can't see anything in the manual that could be relevant.