What are Reddit's favorite Audio Router Apps?
Loopback
PulseAudio
VB-Audio VoiceMeeter
Soundflower
AudioBus
Carla
SoundDesk
Top Reddit reviews mentioning Audio Router Apps:
If you're too lazy to click:
- Automatic routing improvements
- Beamforming and various other new features in the WebRTC echo canceller
- Various improvements in module-role-cork and module-role-ducking
- LFE remixing disabled by default
- memfd-backed shared memory transport
- Support for sample rates up to 384 kHz
- webrtc-audio-processing dependency minimum version bumped to 0.2
- Changed the C standard from C99 to C11.
More details here.
Backup /etc/pulse/default.pa
Open /etc/pulse/default.pa and find the line load-module module-role-corkchange it to #load-module module-role-cork This will stop apps from muting other apps
Then find load-module module-bluetooth-policyand change it to load-module module-bluetooth-policy auto_switch=0
According to https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/#module-bluetooth-policy auto_switch=0 should disable switching between HFP and A2DP modes based on active streams.
Save the file, and reload pulseaudio - pulseaudio -k
Don't record digital audio from your computer by going through an interface, that's silly. Just use software to route stuff around. On Mac use this Soundflower or something equivalent:
https://github.com/mattingalls/Soundflower
There are other options on PC.
> Can't find any documentation on avoid-resampling
I can't imagine why, considering how complete and comprehensive PA's documentation is.
--
avoid-resampling With avoid-resampling = yes, PulseAudio automatically configures the hardware to the sample rate which the application uses, if the hardware supports this sample rate (needs PA 11 or higher)
... says ArchWiki.
That's often the problem. For many things in Linux, the "Advanced Settings" dialog is missing in GUI. Same goes for power settings and touchpad settings. What could be solved with a couple of mouse clicks, needs spending a hour or two reading documentation and dicking around in command line.
Here's an example. I recently noticed that there is some A/V sync discrepancy which could be solved by adding some latency to audio. I found out that the PulseAudio loopback module can be used to add delay. The documentation for that module has been sitting in a browser tab waiting for some free time for me to figure out how everything works. I have no idea how PulseAudio modules are loaded or how and where their parameter syntax is defined, and it all just feels like a chore. There could have just been a simple GUI slider to add some delay.
Once again: the GUI is much more discoverable and much faster way to change settings if you only need to do some one-time thing. The CLI is often praised by nerds, but is faster only if you already know specifically what to type and you have to do it often.
By default the pulseaudio socket is not mounted, you have to request that, and that choice is overridable by the user. Although the permission request and overriding is very poorly exposed in the UI/CLI atm, clearly there is work to do there.
We're working on updating pulseaudio to have more finegrained permissions. See https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer/AccessControl/
Pulseaudio has a Acoustic Echo Cancelling module, called module-echo-cancel. The algorithm cancels echo and also, in effect, filters noise. I use it myself, and it's not perfect but it is an improvement.
In default.pa
load-module module-echo-cancel source_name=<filter out> source_master=<filter in> aec_method=webrtc
Further, you can use an equalizer to emphasize your voice. The difficulty is there is no equalizer module for pulseaudio sources (microphones). There is likely a way to connect source to sink, filter and then go back from sink to source with module-loopback. Or you might be able to use module-jack-source and apply an equalizer to the source with JACK. I've never done these so I can't help on this end.
I had this problem for months on different machines, a MacBook Pro and a Mac Pro. When the simulator runs, it switches your audio config into "Call Mode" by activating your microphone. I have no idea why, but here are two solutions:
- Solution #1: Choose a different microphone. If your using headphones, choose the internal microphone as your input: Open system preferences -> Sound -> Input. This makes the headphones work fine.
- Solution #2: Install Soundflower and same trick. Choose the "virtual" input for your Mac, one of the soundflower channels.
When I started working on the Mac Pro I discovered that it doesn't have an internal microphone, so my old little trick didn't worked anymore. If you're not using headphones and just want the audio coming from your laptop to be good, then solution #2 will do the trick.
As someone who almost went crazy because of this problem, I wish you good luck!
The best thing is to get a very directional mic that sits close to your mouth on a headset and turn down the sensitivity/gain until only your voice registers. When shopping look for the term 'cardioid'
If you have a directional mic (you might already) then make sure the positioning is good. You might want to invest in a boom if it's on your desk.
VoiceMeeter can help if you don't want to upgrade hardware. Setting it up can be a little confusing but there's a bunch of youtube vids on how to do it. Once you get it working you'll want to mess with the Audibility setting to try to filter the keystrokes without messing with your voice. I've tried it out with a decent condenser mic but was never really able to get rid of the keystrokes and ended up just swapping to a modmic set to unidirectional mode.
you can use the ladspa module to load whatever fun plugin you want.
there is also an aur package called pulseaudio-equalizer-ladspa which will setup a ladspa equilizer for you, with a gtk gui.
VoiceMeter is what you are looking for. https://www.vb-audio.com/Voicemeeter/index.htm
You can set different input and output channels and can toggel them. If you need more than 2 channels use the Banana version.
Works good for me.
You can definitely do this with PulseAudio. Here's how I would do it:
- Create a null sink.
- Loopback your microphone to this null sink.
- Create a combine sink that combines your primary sink (sound card, probably) with the null sink.
- Route game audio to this combine sink. It will now play both through your speakers/headset and into the null sink we created.
- Make the monitor of the null sink the input for Discord/Skype. It will record your microphone and any applications routed into the null sink or the combine sink. You may have to do this using
pavucontrolorpavucontrol-qt-- at least in my case the KDE audio applet and the Discord input device selection do not display PulseAudio monitor sources.
Example commands (fill in the correct names for your interfaces):
pacmd load-module module-null-sink sink_name=Virtual
pacmd load-module module-loopback source=MICROPHONE_SOURCE sink=Virtual
pacmd load-module module-combine-sink slaves=Virtual,SOUNDCARD_SINK
To make these stick, put them in ~/.config/pulse/default.pa and then restart Pulse with systemctl --user restart pulseaudio. You can figure your sink and source names with pacmd list-sinks and pacmd list-sources respectively. You can change the name the null sink and combine sink show up in your system audio control interfaces with pacmd update-sink-proplist SINK_NAME device.description="Human readable name here" (if you're invoking this from a shell prompt, you may have to escape the quotes somehow, but how exactly that's done is escaping me right now. In default.pa this will work as is)
No need to listen to your own voice or anything like that, and it won't create an echo for people listening in on the stream because you can do per-application routing in Pulse.
Further reading: https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/
> Audacity for example doesn't work at all with PA.
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/PerfectSetup/ :
> Audacity as packaged in Fedora 11 and 12 works with PulseAudio.
>Audacity has now been packaged with a proper "alsa: pulse" device listed, in a ppa for ubuntu intrepid. See https://launchpad.net/~diwic/+archive
You should get Audiobus on the iPad and use the iPhone for Audiobus Remote: https://audiob.us/remote
This app is great. It puts controls for all your (compatible) music apps on the iPhone screen. So you can have the main app you're playing in up on the iPad screen, and then have controls for all the apps running in the background showing up on your iPhone.
I love Samplr and it's a really unique tool, but it needs better support. I'm not sure you really play a synth into it like that either. You could play the synth into another app (like garageband) and then, using Audiobus, record that sound into Samplr.
So yeah, check out Audiobus and the companion app Audiobus Remote. I think they come in a bundle and they're so worth it. They really expand what you can do with an iPad.
If someone actually has it working I would like to hear about it. It was planned for PulseAudio 13 but then for some reason was not. If you look here it is crossed out in the release notes: https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/13.0/ No newer versions since 13 has mentioned it again.
I don't think there's any way to do this in-game; you can't even select the audio output, let alone different ones for different purposes.
One option would be to run two copies of the game: one with music on and sound effects off, the other the opposite. Then it should be possible to route one application to one output, the other to a different one. At least it should be possible with third party software. A quick Google brought up an article about Audio Router for Windows 10. And on macOS I can verify that Loopback does the job - though it's not free. I'm sure Linux has various free options.
Or, easier, what about doing a one-time recording of X hours of the music, then using that recorded music as background in all future projects? Adding it to your project when editing pre-recorded stuff, or, if live-streaming, just playing it in the background with any media player.
I don't believe the Factorio music is event-dependent? I know it has main tracks and 'interludes/world ambience' that plays in-between. But I don't think that depends on what's actually happening in the game? So I would think that capturing the music once, over a suitably long duration, then playing that back - perhaps starting at a random point each time - would be sufficient to replicate the in-game experience fairly well?
I don't know for sure as I haven't listed to the music for a while, but that's certainly what I'd try first.
Similar solution to u/clocksinbox, but I recommend using Voicemeeter or Voicemeeter Banana to make XIV play nicer with your audio devices. It's a virtual audio mixer that's very flexible and has a variety of other uses as well (echoing audio output from specific programs, for example).
You can find it here, including Banana and Potato (you probably won't need all the channels in Potato though).
Setup guide: https://imgur.com/a/ojKeb
That doesn't look like sandboxing.
> How to pass the sound in my regular system?
> How to access files located in my regular system?
Bind mounts.
> Where would I place a config file?
The usual location.
When I had a related issue I was able to fix it by adjusting /etc/pulse/default.pa based on what I read from the PulseAudio modules page. Check out the sections for module-switch-on-port-available and module-switch-on-connect. Hope that helps!
edit: I just checked my /etc/pulse/default.pa and see that there is a pacnew file alongside it which removes load-module module-rescue-streams so might be worth checking if that is related.
I'm going to screw something up so someone smarter will correct me, but PulseAudio is "a sound system for POSIX OSes, meaning that it is a proxy for your sound applications" ^[0] . So if something is wrong, you get no sound and since there is no text in the comic it's implied there's an issue and I'm just kind of rambling at this point, did that help?
I was interested so I looked it up...
From: https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/11.0/
"Now it's possible to set "avoid-resampling = yes" in daemon.conf, which will make PulseAudio configure the hardware to whatever rate the application uses (if the hardware doesn't support the application's rate, resampling is of course still done)."
https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/BrokenDrivers/
> And yet pulse pops and alsa doesn't. I guess it's my fault for not knowing enough about sound drivers to google "timer-based scheduling" beforehand.
.....
By default Pulseaudio is a per-user thing that is invoked upon login, so it's the sound of it starting up and grabbing an audio output. You can change your PA config to be systemwide, but there are considerations (linked in article).
Yes, there are plenty of options out there. Several Screen recording apps come with their own implementation, for example. All of them have in common to essentially add the system audio output as a pseudo-device.
Other than that, you could try Loopback:
So, currently the automated setup is quite simple because Instagram removed the ability to auto-magically upload content to accounts unless the account is a business account. Considering that my account is a "creator" account, I'm stuck with doing things manually.
However, from Max I currently press a toggle that starts a countdown, and when the countdown goes to 0, the 60 seconds start. The toggle triggers a bash script that contains a few lines to automate the recording, trimming and cropping of the video. Core elements of this chain are:
- screencapture (leveraging macOS' internal screen capture utility that is now available from command line). -V66 to record only 66 seconds of video.
- ffmpeg (to do all the video trimming and cropping). -ss to trim the video to exactly 60 seconds, -filter:v "crop" to crop it to an exact square, -af "afade" to do fade in/out and avoid clicks.
I also use "SwitchAudioSource" from switchaudio-osx to auto switch the internal settings of my audio routing. I am currently using a free version of LoopBack to make things easy but you could do it with any other re-routing software.
https://rogueamoeba.com/loopback/
Loop back audio. It's an digital internal audio router. Free download but you can record only 20 min of highquality audio. After that you need to pay for the whole think. So far 20 min has been enough to get the sample I need
Yes, I have Roland TD-17, which can be used as audio interface; I'm using Mac.
Regarding the microphone, I'm just using build-in mic when we discuss something. In order to it, you have to switch input device from you audio interface to internal microphone.
Another solution (for Mac) would be using a loopback software, which let you create a virtual mixer, in which you can enable/disable your mic with toggle button. Another benefit of using loopback is that it can route sound from your VST (EZDrummer, Addictive Drums, Superior Drummer, etc) to the given virtual device. This is very handy in case an application doesn't support it natively (Jamulus). I think SonoBus has support of VST, but I haven't tried it yet.
Happy jamming!
A slightly less overkill way to use analyzer plugins that doesn't involve firing up a whole full on DAW (Ardour) would be to simply use a stand alone plugin host like Carla. http://kxstudio.linuxaudio.org/Applications:Carla
There is a blurb on their page, but it is mistakenly dated 2014.
https://github.com/mattingalls/Soundflower/releases
Apple makes you jump through a few hoops. The first time you run the installer (Soundflower.pkg), it will ask for your admin password, and will FAIL! A security alert will appear, with a button to take you to System Preferences "Security & Privacy - General" pane. Once there, there should be an "Allow" button (**) that you will need to click on to give permission to use Soundflower (developer: MATT INGALLS). Then, RUN THE INSTALLER AGAIN. It should inform you installation was successful. If the "Allow" button is disabled, you may need to click the lock icon in the bottom lower left corner first.
(**) If you see an "Open Anyway" button in the Security Preferences, this is something different!!! Most likely because you tried (and failed) opening the installer by double clicking without holding down the control key. If so, click the "Open Anyway" button which will display another window. Then click the "Open" button in that window to launch the installer. Now you can follow the instructions above to get the "Allow" button to appear in the Security Preferences.
$200 will get you into a very good basic interface and standard dynamic mic that will make just about every issue you are having disappear.
But nobody listens to Zathras.
Look at VoiceMeeter and record into a single computer.
https://www.vb-audio.com/Voicemeeter/index.htm
It will help some assuming you can get two Yetis to work on a single computer without returning one to Blue to have the USB GUID changed per point three on their FAQ.
https://www.bluedesigns.com/faq/
This is a road of pain and fear and suffering leading to anger and hate but it seems necessary to bring balance to some podcasts.
!
You will be looking for a long time. The short answer is you need a hardware or software stereo mixer. Or maybe wire the analog to your sound card or get something like a Behringer UCA202.
You need four inputs like the Behringer UMC404HD instead of the 2i2 or a basic stereo mixer to do this easily.
Edit: I suspect your first stop will be VoiceMeeter followed by a bottle of acetaminophen.
https://www.vb-audio.com/Voicemeeter/index.htm
!
Reaper, like many DAWs, can only work with one audio interface at a time. If you are on Windows, try VoiceMeeter which lets you combine multiple hardware interfaces into a virtual mixer.
EDIT: VoiceMeeter is donation ware, so free if you want it to be.
Most programs/games/etc will have the option to select which devices for Input/Output instead of the default (which you can change in Windows' audio management).
I can also recommend VoiceMeeter from VB-Audio (link https://www.vb-audio.com/Voicemeeter/index.htm) for more advanced use of your devices
I use the program Voicemeeter. It allows me to bypass the normal inability to use 2 USB mics with Audacity (And presumably any other audio program you'd like to use)
The same company which makes the tool you're using also has a more advanced tool that would do exactly what you want. Actually, they have two, one relatively simple, one more complex (even supports sending audio between computers over a network). Both are just as free the virtual cables. https://www.vb-audio.com/Voicemeeter/index.htm
You need a mixer of some sort to mix in (get it?) other audio, either a physical mixer or a virtual one.
On Windows, look at VoiceMeeter.
https://www.vb-audio.com/Voicemeeter/index.htm
There are a few other solutions. It also depends on your existing hardware.
!
How do you start qutebrowser? As a gui applications I assume not through your init system ;). DE autostart? display manager startup files? WM config files?
What if you manually start pulseaudo before your start qutebrowser with pulseaudio --daemonize? (More details here)
I used to use a Windows build of pulseaudio for this like a decade ago for a similar situation (streaming from mpd, although I didn’t use zeroconf). There is a build available here but as you can see it’s just a bit out of date :p
Since these kind of things are normally done in driver/software not inherently no.
You can however apply your own software equalizers/surround emulations. See pulseeffects/easyeffects for an extensive catch-all soution or e.g. on pulseaudio there's the virtual surround module
>Creative Sound Blaster X4
Yea... creative are jerks to Linux... Do not expect drivers on Linux.
https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/BrokenDrivers/
> Regarding snd-emu*: Creative doesn't like Open Source -- there are no docs available. If you buy Creative it is hence a bit your own fault.
Check ArchWiki's PulseAudio entry, look at the related pages, if there is nothing in the main article.
$ less /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf
Read through the top and bottom comment sections, it should give you an idea what the configuration options are. Create a new file in that directory with a new profile definition that includes channel mapping as desired. Not sure if using the environment variable is the only way to load the profile as per documentation.
https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/Profiles/
Iit's technically possible using pulseaudio however the configuration isn't trivial (on the cleint side any player that can connect to a multicast RTP stream will work). I'm not even sure if it's actually possible in newer libreelec as I'm not sure if they still use pulseaudio or have all the modules available.
It's likely much easier to use the pi's line-out and either an analog mixer or input that in to your computer and mix it digitally. If that's too low quality you could also split out the hdmi digital audio in to the computer, though digital-in audio isn't too common on computers.
Download, compile and install latest Pulseaudio (pulseaudio 15.0-25-g19ad).
Version 15 includes support for Bluetooth A2DP AVRCP Absolute Volume.
I hope it solves your issues.
I don't do much with audio and multimedia but I can't use Pipewire until the echo-cancellation-module is available. My mic sounds awful without it.
On the other end I look forward to manage my audio streams with carla :D
I'm not versed with pulseaudios modules, but there seems to be a module that should fit your needs exactly https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/#module-switch-on-port-available
Instead of making a shell script and autostarting it, you can put the commands in ~/.config/pulse/default.pa. Just remove the "pactl" from the beginning of the commands.
Here's what I have in mine (my concern is keeping voip out of recordings/streams rather than music, which is why I call one of these "VoIP Output", but the principle is the same):
load-module module-remap-sink sink_name=default_output master=alsa_output.usb-BEHRINGER_UMC204HD_192k-00.analog-surround-40 channels=2 master_channel_map=front-left,front-right channel_map=front-left,front-right remix=no update-sink-proplist default_output device.description="Default Output" load-module module-remap-sink sink_name=voip_output master=alsa_output.usb-BEHRINGER_UMC204HD_192k-00.analog-surround-40 channels=2 master_channel_map=front-left,front-right channel_map=front-left,front-right remix=no update-sink-proplist voip_output device.description="VoIP Output"
I use remap sinks instead of loopbacked null sinks, which lets me do the same thing with fewer modules. I also use update-sink-proplist to give the sinks a nice human-readable name in pavucontrol and the KDE audio switcher applet.
EDIT: and you can indeed set these sinks as the default "device" using audio switcher applets, but failing that I believe pactl set-default-sink can do the same thing.
I have a scarlett solo and am on Arch, but I'm still using pulseaudio and pavucontrol so I wouldn't imagine it's too different. On pavucontrol make sure the 2i2 is set to the fallback device on the "Output Devices" tab, and on the "Configuration" tab I have every other device turned off, and the scarlett solo set to the "Analog Stereo Duplex" profile (I imagine due to the 2i2 having 4 channels it might have a different option, but look for the Analog ... Duplex option). You can read more about pulseaudio default/fallback devices here: https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/DefaultDevice/
pulseaudio has a module for that - module-null-sink
use this command to add the module to pulseaudio
pactl load-module module-null-sink
and to make it the default sink
pactl set-default-sink null
to listen to an audio stream, go to pavucontrol --tab=1 and click on the drop-down menu to re-direct audio stream in from null to the audible output sink
I do believe you CAN run Pulseaudio on Windows. It's a pretty old version, but it should work. With that you could create a bridged network (so you can see both the host and the guest on the network) and use Pulseaudios ability to send audio over a network to send it to your host.
Here is Pulseaudio on Windows https://www.freedesktop.org/wiki/Software/PulseAudio/Ports/Windows/Support/
As I understand it no its not possible. Here is some documentation about the problem and designs to solve it: https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/Developer/AccessControl/
It seems this will never be solved and we'll have to wait for Pipewire
For those that don't know, you can also change pulseaudio's resampling method to be lighter on cpu, or higher, if you need. https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Audiophile/
My default.pa is default one provided by distribution. However, thank you very much, once I knew that this is intented behaviour and not my mistake, I was able to find https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/#index28h3, so changing load-module module-stream-restore to load-module module-stream-restore restore_device=false did the trick.
I really don't understand why restore_device defaults to true. I mean, restoring volume and mute state can be useful, but don't tell me this is something majority of users find useful.
So, thanks again :)
Iirc, PulseAudio itself still has a ducking module built in, but the VOIP audio stream has to report itself as a different role from your music: https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/#index61h3
I've had issues along these lines in the past. On most Linux distributions, we use PulseAudio as an audio manager. Pulse cones with a collection of loadable modules, and if I remember correctly there is a module that measures and significantly reduces background noise. I don't have a precise solution, but you can find documentation for PulseAudio at https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/
I hope this helps you find a solution.
There's a PulseAudio server (i.e. the part that ouputs audio) for Windows here.
You should be able to do this entirely with Pulseaudio with a setup like the following. This assumes that both guest and host have their own physical audio output devices and you're simply switching outputs on your receiver/amplifier If this isn't the case for you, let me know how your audio devices are connected and I can provide alternate instructions.
- Host runs a PA server
- Guest runs a PA server
- Load module-tunnel-sink on the host to create a new sink (audio output) that sends audio to the guest PA server
- Load module-combine-sink on the host to set up a new sink that sends its output to both the host's actual audio output, and to the tunnel sink created above.
- Make the combined sink the default sink. This will cause the host's audio to be output on both it's own audio device, and sent to the guest
- Qemu outputs guest audio as a client to host PA server. Explicitly send it to the host's real audio output, not the combined sink, or you'll have created a loop.
- In the Windows VM, use VoiceMeeter to create a virtual audio output device that sends to both the VM's emulated sound card and to the passed-through soundcard, and set this virtual device as the default playback device.
It can be easily done by adding a Pulse sink.
Documentation here. You need a HRIR filter specifically calibrated for your head for best results. But a bunch are linked there.
I don't know exactly what AudioFX sounds like but you will probably not replicate it. Professionally made virtual surround typically also has some layers of equalization and matrix decoding filters.
> planning to do initial boots on linux with a later install of windows (potentially dual booting).
well, you build will be a nightmare on linux. Creative and a Nvidia card. ouch.
https://www.freedesktop.org/wiki/Software/PulseAudio/Backends/ALSA/BrokenDrivers/
> Regarding snd-emu*: Creative doesn't like Open Source -- there are no docs available. If you buy Creative it is hence a bit your own fault.
This sounds like something that could be implemented in the pulseaudio config without the need to look at MPRIS or writing an extension. The role ducking module should offer everything you need:
https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/#index66h3
Here is a list of the possible roles:
All of your use cases can be done in Pulse?
> routing audio between clients
Pulse Audio has pipes and sinks so it could be done to reroute the audio? - https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/#index2h3
> realtime DSP (Here is instructions on using compression with Pulseaudio
https://www.bfccomputing.com/dynamic-range-compression-for-pulseaudio/
I believe this should be possible with Audiobus. Use jack as an input and then route it to an app that allows to pass the input to Airplay speaker. Quick search of the compatible apps showed this app: multiroom-music-radio-player
You can ask on Audiobus forums to confirm the setup will work before you spend money on it.
Samplr, Fugue Machine, Xynthesizr, Borderlands, Impactor, Sektor, Diode-108, SP-Electro.
Tools: Audiobus, Audioshare, MusicIO, Touchosc and iLemur.
If you're on a Mac, you can use a program like Loopback to route computer audio into a DAW and then set your mix up on a separate input. You could monitor on headphones to capture only your voice on the mic.
Yep, was about to say... this has been posted about ad nauseam. I've been using NDI tools since mid 2020 for remote sessions. It turns your NLE output into a webcam that you can select in Zoom. You can couple that with something like Loopback to ensure you're routing the proper audio channels.
Evercast basically ripped off OBS and started charging $600/mo. It's insane.
You need to be able to pass the audio into Discord as a source, likely alongside your mic, which means creating some kind of aggregate virtual audio device / audio loopback / passthrough. There might be better search terms but I unfortunately don't know them.
I'm on Mac so I use Loopback - not sure what specific solutions work well on Windows.
eta: oh, I'm also aware that there is a Syrinscape subscription tier that allows something like this. You generate a link and invite your players. But you're locked into their sounds and I'm not sure if all players need the software, too.
I do this on my Mac with an audio app called Loopback: https://rogueamoeba.com/loopback/
It’s pricey but allows you to, for example, create a virtual audio device that route ableton inputs 3+4 along with your mic to a single input which you can select in discord.
Similarly, you can route your discord inbound and ableton cue so you can monitor whatever you’re playing and still hear people talking (without feeding their audio back into the stream)
Probably. I would imagine it would be alot easier with macOS. If you are running mac this Should be useful. Otherwise you can use virtual ausio cable if you are on windows. Good luck.
https://rogueamoeba.com/loopback/ creates loopback devices like Soundflower and can also intercept application audio and remap or duplicate inputs and outputs. It's $99, but Rogue Amoeba are long-standing reputable developers.
Thanks for the mention!
OP, for this, you'd actually want our app Loopback (https://rogueamoeba.com/loopback). You'd create a virtual audio device that has Chrome as its source, then set that as the input in Logic. This would get the audio from Chrome directly into Logic.
Had the same using my sound system when hooked up via optical audio on my Hackintosh.
Use https://rogueamoeba.com/loopback/. This allows you to setup a virtual sound card which can be controlled like normal and set the system audio as input and your interface as only output.
Regarding sampling from the computer I'd highly recommend this https://rogueamoeba.com/loopback/
In regards to the cable for your phone, the MK3 supports stereo 1/4" TRS cables. So you'll need a male 3.5mm to male stereo 1/4" TRS cable.
PSA: This is tangentially applicable to this situation, but for those of you with multiple USB mics:
If you are on OSX and are willing to part with $99 you can use Rogue Amoeba's Loopback program to support multiple USB mics at a time. I recently combined my audio interface with 2 XLR mics and 3 concurrent USB mics (a yeti, snowball, and meteor) as a composite audio device that I used in my DAW. Worked great.
Here is how it looks (the mics are disconnected at the moment): http://imgur.com/a/BOOyx
I am not affiliated with Rogue Amoeba, just a fan.
You need a virtual "audio cable" software for that. I do not have any experience with Mac at all, but people seem to like Loopback for that purpose: https://rogueamoeba.com/loopback/
Basically you create multiple virtual audio outputs and assign each to a application (e.g. one for the game, one for iTunes, etc) and then route them to multiple outputs which you can then mix as you like, e.g. output 1 only game, output 2 game plus iTunes, etc and select the one you like in OBS as desktop audio source or aux source.
Can't speak for other operating systems, but if you're on a Mac, I would use Soundflower. It lets you route the computer's system audio into your DAW input, where you can do whatever processing you want to it. Let me know if you have any questions!
This worked. Some ancient versions dominate the search results; as best I can tell the most current version is here: https://github.com/mattingalls/Soundflower and it works with Catalina.
I just realized this doesnt record sound. This might do it for the sound... freeware that should work... Just scan just to make sure but it looks good. https://github.com/mattingalls/Soundflower/releases/tag/2.0b2
The issue with this is that it doesn’t record any sound that is output by the computer. To get around this, there are a few virtual audio output devices such as Soundflower. By selecting “Soundflower (2ch)” (or whatever alternate virtual output device is being used) as the microphone while screen recording, the audio output will be recorded also.
I assume you do not have access to an external sound card.
The common approach is to use Soundflower to route system sound into a virtual "microphone" that can be used by Audio Analysis.
Soundflower is a small free app that creates virtual inputs and outputs in System Sound. They also show up in Audio Analysis as inputs. The trick is to route the system sound into Soundflower, making you able to playback the music on your iMac's speakers as well as getting a clean input into Audio Analysis.
Install Soundflower, and install SoundflowerBed as well. SFB is not necessary, it's just a nifty GUI for the menu bar that I prefer to use (you'll read why below). You may need to restart the system post-install.6
In System --> Sound, set your Output to the new "Soundflower (2ch)". Repeat for Input, also setting it to use "Soundflower (2ch)".
What is happening is that all system sound (music, app sounds etc) will play through the virtual Soundflower (2ch) output. And instead of using the default microphone input, we get clean sound from the source, as the output is routed directly into the input.
The downside is, no sound will play through the internal speakers with this setup. The sound playing through non-existing speakers (and recorded through a non existing microphone). My workaround is to use SFB to simultaneously route the Soundflower output through the speakers. Run SFB, click the icon on the menu bar and choose and output from the drop-down menu. It's probably something like "Internal Speakers".
Well, the old version isn't compatible, but the one on Matt Ingall's GitHub does. Download here.
That one doesn't include Soundflowerbed anymore, but there is a Soundflowerbed 2.0 fork that works.
Soundflowerbed isn't strictly necessary, as long as you're routing through a DAW, though - you can just create an Aggregate Device in Audio MIDI Setup to achieve the same.
Yup, Quicktime won't record your Mac's audio (by default).
If you want to use Quicktime, I use Soundflower. It turns your Mac's audio output into an input. You can then select the sound input in Quicktime when doing an audio or video recording.
If you have an optical drive in your laptop/workstation and you have an audio interface that supports loopback, you can just crank up ableton, enable the loopback ouputs in preferences, change the master output from 1/2 to 3/4 (so you don't feed back), then create a new audio track that takes input from your loopback outputs. Then you just hit play on the DVD at the scene you want to record, and start recording on the channel you created in ableton. If you don't have an audio interface or don't have one that supports loopback, you can use something like Soundflower or Loopback to accomplish the same thing.
Here you go https://github.com/mattingalls/Soundflower/releases/tag/2.0b2
This link was found on the OBS forums, can't really find it from Google i guess.
I don't know if you need the Blue Sherpa software. It may help.
https://www.bluedesigns.com/products/sherpa/
Also look at Voicemeter.
https://www.vb-audio.com/Voicemeeter/index.htm
And as always, multiple USB mics are a path of pain and anger and suffering.
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Look at Craig.
You need a mixer of some sort For a virtual mixer on Windows look at Voicemeeter.
https://www.vb-audio.com/Voicemeeter/index.htm
There are other, likely better ways with hardware but these won't work with the Snowball.
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I really don't have any good points for comparison as the only other things I've plugged my K701s into are a Denon AVR5700 and my old SB0460 with LiveDrive.
I can't hear interference from the CPU and other components anymore, and provided I'm listening to something with high production value, I can't really find complaint. Granted, I'm not feeding something as demanding as planar cans...
Because of the relatively unique setup, I've been using VoiceMeeter as my primary sound device to manage all the audio I/O...
Eu mestro via Discord, usando o Voicemeeter e Virtual Audio Cables para fazer com que o som que toco no iTunes seja ouvido pelos participantes (porém sem eles ouvirem o retorno deles mesmos). É meio chato para configurar (recomendo deixar tudo anotado e olhar alguns tutoriais), mas funciona muito bem.
Yes. You can use VoiceMeeter, route your audio through it, and then select it as the audio device in Ableton. Make an audio track, select 'ext. in' as the input source, select '1/2' so it's stereo and hit record.
I've only used this with OBS and for streaming but it should function the same with Shadowplay since you can select what is the main audio output for the recording.
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A combination of Virtual Audio Cable and Voicemeeter from VB-Audio. They have a pretty good video tutorial on how to set this up. I used it to exclude my Discord and Mic input from stream unless I used 'pushed-to-talk' to allow my mic to pass through to stream.
I was wrong, It's called voicemeeter. https://www.vb-audio.com/Voicemeeter/index.htm + https://www.vb-audio.com/Cable/index.htm needed for this to work.
Route your audio to voicemeeter and then route voicemeeter back to windows.
If you have Windows 10 you can use this one: Speak+ on the Microsoft Store. I like it because you can just press enter to say the message and you can also save messages. Then use Voicemeeter to route the audio to your microphone.
Multiple USB mics are a road of pain and suffering and anger.
Having said that, it can be done but it depends on the OS you are using and a few other bits.
Also, if you are using two Blue mics, per point three in the Blue FAQ you may have to mail one back to the factory to have the USB GUID modified.
https://www.bluedesigns.com/faq/
But if you are using Windows, look at VoiceMeeter or VoiceMeeter Banana.
https://www.vb-audio.com/Voicemeeter/index.htm
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I have several options in a post that AutoMod ate because I mentioned Spreaker Studio and AUtomod hates Spreaker plus a few other options. It's not a great idea and Aggregate Devices are unstable.
For Linux, Jack comes to mind and Windows would look to Voicemeeter, though this doesn't apply to OP.
https://www.vb-audio.com/Voicemeeter/index.htm
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Go to the Windows "Sound" settings, select the "Recording" tab and Properties on your recoring device.
There you go to the "Listen" tab and check "Listen to this device". You can also select the output device if you don't want it to be the default one.
To split the output before outputting to your hardware (e.g. for OBS recording) you can look over here:
https://www.vb-audio.com/Voicemeeter/index.htm
or
https://www.vb-audio.com/Voicemeeter/banana.htm
I haven't tested it but should be doable.
How are your speakers/headset setup? If you have a USB headset you use for in game audio as well as a microphone then you can connect the tablet using a TRRS to TRS splitter to the onboard 3.5mm connectors and use something like VoiceMeeter to split/combine the audio (I have a similar setup on my PC with a lot of backend mixing that "just works" that I don't quite understand why it works at all.)
Without downloading extra software, no. Windows does not support that natively (for some weird reason).
You can download VB-Audio VoiceMeeter for free and set it up that way. After installing it, you'll likely need to reboot your computer. Then just start it up and then select the two outputs you want the sound to go to (the box in the last quarter of the interface) by clicking the A1 and A2 buttons at the top and then assigning the outputs. Change your audio output device in Windows to be the VoiceMeeter Input and you're good to go!
Look up a free program called "VoiceMeeter". (Or you can used the advanced version called "Banana".)
It works as a mid-way point for two usb inputs (i.e. headset mics) to be mixed into a single channel which can then be recorded by Audacity or whatever you're using. Fair warning - it records it as a single track, so you can't go back in post and edit one person's voice independently from the other, so make sure to do a trial run just to test and balance individual levels.
Other than that, yeah, Blue Yeti is a solid investment that has multiple recording arrangements built in. You can toggle whether you want to record all around (which is easiest to do for recording multiple people, but can pick up a lot of room noise), or just in front of the mic, or focusing on the front and back at the same time while eliminating the sounds to the sides. Usually runs $100 - $150, depending on what style and bundle you choose.
Welcome to an intentional design feature in Windows that's been around since forever. Specifically that Windows can only have one playback device by default.
The solution in your case is to run Voicemeeter on your gaming PC so audio gets sent to both your regular soundcard and HDMI video at the same time. You install Voicemeeter, set Voicemeeter as the default sound input/output, then select your audio routing inside Voicemeeter. Don't forget to set Voicemeeter to automatically start every boot.
https://www.vb-audio.com/Voicemeeter/index.htm
Plus is really is a case where you'll need to RTFM:
https://www.vb-audio.com/Voicemeeter/Voicemeeter_UserManual.pdf
You can use Voicemeeter. The free version will work. You set one input to your actual microphone and the other input to one of the virtual outputs it adds. Then set the output of whatever application will be playing to one of the virtual output devices. Lastly, make make your default input device one of the virtual input devices.