This post is largely superseded by a new post on the same topic. The new post includes more step-by-step setup instructions for an audio-focused Linux distribution, tips for how to use digital audio workstation tools to manage multiple microphones, and hardware recommendations for podcasting use.
The Crossbox Podcast is going upmarket: I now have two USB microphones, and for the March episode, parvusimperator and I will each have one directly in front of us. This is a wonderful advance for audio quality, but it does pose some problems:
- Audacity, our usual recording tool of choice (and probably yours, if you ended up here), only supports recording from one source at once.
- Though other tools support recording from multiple sources, the minor variations in internal clocks between two USB microphones mean that each microphone has a sample rate which varies in a slightly different fashion, and that longer recordings will therefore be out of sync.
Modern Linux, fortunately, can help us out here. We have need of several components. First, obviously, we need two microphones. I have a Blue Snowball and a CAD Audio U37, with which I’ve tested this procedure1. Second, we need a computer with at least two USB ports. Third, we need the snd-aloop kernel module. (If your Linux has ALSA, you probably already have this.) Fourth, we need JACK, the Linux low-latency audio server. Fifth, we need the QJackCtl program.
Before I describe what we’re going to do, I ought to provide a quick refresher in Linux audio infrastructure. If you use Ubuntu or Mint, or most other common distributions, there are two layers to your system’s audio. Closest to the hardware is ALSA, the kernel-level Advanced Linux Sound Architecture. It handles interacting with your sound card, and provides an API to user-level applications. The most common user-level application is the PulseAudio server, which provides many of the capabilities you think of as part of your sound system, such as volume per application and the ‘sound’ control panel in your Linux flavor of choice. (Unless you don’t use Pulse.)
JACK is a low-latency audio server; that is, a user-level application in the same vein as Pulse. It has fewer easily accessible features, but allows us to do some fancy footwork in how we connect inputs to outputs.
Now that you have the background, here’s what we’re going to do to connect two mono USB microphones to one computer, then send them to one two-channel ALSA device, then record in Audacity. These instructions should work for any modern Linux flavor. Depending on the particulars of your system, you may even be able to set up real-time monitoring.
- Create an ALSA loopback device using the snd-aloop kernel module.
- Install JACK.
- Build QJackCtl, a little application used to control JACK. (This step is optional, but makes things much easier; I won’t be providing the how-to for using the command line.)
- Use JACK’s alsa_in and alsa_out clients to give JACK access to the microphones and the loopback device.
- Use QJackCtl to connect the devices so that we can record both microphones at once.
We’ll also look at some extended and improved uses, including some potential fixes for real-time monitoring.
Create an ALSA loopback device
The ALSA loopback device is a feature of the kernel module snd-aloop. All you need to do is
# modprobe snd-aloop and you’re good to go. Verify that the loopback device is present by checking for it in the output of
The loopback device is very straightforward: any input to a certain loopback device will be available as output on a different loopback device. ALSA devices are named by a type string (such as ‘hw’), followed by a colon, then a name or number identifying the audio card, a comma, and the device number inside the card. Optionally, there may be another comma and a subdevice number. Let’s take a look at some examples.
hw:1,0: a hardware device, card ID 1, device ID 0.
hw:Loopback,1,3: a hardware device, card name Loopback, device ID 1, sub-device ID 3.
For the loopback device, anything input to device ID 1 and a given sub-device ID n (that is, hw:Loopback,1,n) will be available as output on hw:Loopback,0,n, and vice versa. This will be important later.
You should be able to find JACK in your package manager2, along with Jack Rack. In Ubuntu and derivatives, the package names are ‘jackd’ and ‘jack-rack’.
QJackCtl is a Qt5 application. To build it, you’ll need qt5 and some assorted libraries and header packages. I run Linux Mint; this is the set I had to install.
Once you’ve installed those, unpack the QJackCtl archive in its own directory, and run
make in that directory. The output to
configure will tell you if you can’t continue, and should offer some guidance on what you’re missing. Once you’ve successfully built the application, run
make install as root.
qjackctl from a terminal. We should take note of one feature in particular in the status window. With JACK stopped, you’ll notice a green zero, followed by another zero in parentheses, beneath the ‘Stopped’ label. This is the XRUN counter, which counts up whenever JACK doesn’t have time to finish a task inside its latency settings.
Speaking of, open the settings window. Front and center, you’ll see three settings: sample rate, frames per period, and periods per buffer. Taken together, these settings control latency. You’ll probably want to set the sample rate to 48000, 48 kHz; that’s the standard for USB microphones, and saves some CPU time. For the moment, set frames per period to 4096 and periods per buffer to 2. These are safe settings, in my experience. We’ll start there and (maybe) reduce latency later.
Close the settings window and press the ‘Start’ button in QJackCtl. After a moment or two, JACK will start. Verify that it’s running without generating any XRUN notifications. If it is generating XRUNs, skip down to here and try some of the latency-reduction tips, then come back when you’re done.
Use JACK’s alsa_in and alsa_out clients to let JACK access devices
Now we begin to put everything together. As you’ll recall, our goal is to take our two (mono) microphones and link them together into one ALSA device. We’ll first use the alsa_in client to create JACK devices for our two microphones. The alsa_in client solves problem #2 for us: its whole raison d’être is to allow us to use several ALSA devices at once which may differ in sample rate or clock drift.
Now, it’s time to plug in your microphones. Do so, and run
arecord -l. You’ll see output something like this.
$ arecord -l **** List of CAPTURE Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC295 Analog [ALC295 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Audio [CAD Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Snowball [Blue Snowball], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0
This lists all the currently available capture hardware devices plugged into your system. Besides the first entry, the integrated microphone on my laptop, I have hw:1 or hw:Audio, the CAD Audio U37, and hw:2 or hw:Snowball, the Blue Snowball.
Next, set up alsa_in clients so JACK can access the microphones.
$ alsa_in -j snow -d hw:Snowball -c 1 -p 4096 -n 2 & $ alsa_in -j cad -d hw:Audio -c 1 -p 4096 -n 2 &
Let’s go through the options. -j defines the label JACK will use for the microphone; make it something descriptive. -d declares which ALSA device JACK will open. -c declares the number of channels JACK will attempt to open.
On to the last two options: like the JACK settings above, -p defines the number of frames per period, and -n defines the number of periods per buffer. The documentation for alsa_in suggests that the total frames per buffer (frames per period multiplied by period) should be greater than or equal to JACK’s total frames per buffer.
Next, set up an alsa_out client for the ALSA loopback device.
$ alsa_out -j loop -d hw:Loopback,1,0 -p 4096 -n 2 &
The arguments here are the same as the arguments above.
Use QJackCtl to hook everything up
Now, we’re almost done. Go back to QJackCtl and open the Connect window. You should see a list of inputs on the left and a list of outputs on the right. Your inputs should include your two microphones, with the names you provided in your -j arguments. Your outputs should include system, which is linked to your system’s audio output, and an output named ‘loop’, the ALSA loopback device.
Assuming you have mono microphones, what you want to do is this: expand a microphone and highlight its input channel. Then, highlight the system output and hit ‘connect’ at the bottom of the window. This will connect the input channel to the left and right channels of your system audio output. At this point, you should be able to hear the microphone input through your system audio output. (I would recommend headphones.) The latency will be very high, but we’ll see about correcting that later.
If the audio output contains unexpected buzzing or clicking, your computer can’t keep up with the latency settings you have selected3. Skip ahead to the latency reduction settings. That said, your system should be able to keep up with the 4096/2 settings; they’re something of a worst-case scenario.
If the audio output is good, disconnect the microphones from the system output. Then, connect one microphone’s input to loop’s left channel, and one microphone input to loop’s right channel. Open Audacity, set the recording input to Loopback,04, and start recording. You should see audio from your microphones coming in on the left and right channel. Once you’re finished recording, you can split the stereo track into two mono tracks for individual editing, and there you have it: two USB microphones plugged directly into your computer, recording as one.
Recording more than two channels
Using Jack Rack, you can record as many channels as your hardware allows. Open Jack Rack, and using the ‘Channels’ menu item under the ‘Rack’ menu, set the number of channels you would like to record. In QJackCtl’s connections window, there will be a jackrack device with the appropriate number of I/O channels.
In Audacity, you can change the recording mode from ALSA to JACK, then select the jackrack device, setting the channel count to the correct number. When you record, you will record that many channels.
Jack Rack is, as the name suggests, an effects rack. You can download LADSPA plugins to apply various effects to your inputs and outputs. An amplifier, for instance, would give you volume control per input, which is useful in multi-microphone situations.
Reduce frames per period
If you’re satisfied with recording-only, or if you have some other means of monitoring, you can stop reading here. If, like me, you want to monitoring through your new Linux digital audio workstation, read on.
The first step is to start reducing the frames per period setting in JACK, and correspondingly in the alsa_in and alsa_out devices. If you can get down to 512 frames/2 periods without JACK xruns, you can probably call it a day. Note that Linux is a little casual with IRQ assignments and other latency-impacting decisions; what works one day may not work the next.
You can also try using lower frames per period settings, and higher periods per buffer settings, like 256/3 or 512/3. This may work for you, but didn’t work for me.
If you come to an acceptable monitoring latency, congratulations! You’re finished. If not, read on.
Fixing latency problems
Below, I provide three potential latency-reducing tactics, in increasing order of difficulty. At the bottom of the article, just above the footnotes, is an all-in-one solution which sacrifices a bit of convenience for a great deal of ease of use. My recommendation, if you’ve made it this far, is that you skip the three potential tactics and go to the one which definitely will work.
Further latency reduction: run JACK in realtime mode
If JACK is installed, run
sudo dpkg-reconfigure -p high jackd (or dpkg-reconfigure as root).
Verify that this created or updated the /etc/security/limits.d/audio.conf file. It should have lines granting the audio group (
@audio) permission to run programs at real-time priorities up to 95, and lock an unlimited amount of memory. Reboot, set JACK to use realtime mode in QJackCtl’s setup panel, and start JACK. Try reducing your latency settings again, and see what happens.
Further latency reduction: enable threaded IRQs
Threaded IRQs are a Linux kernel feature which help deliver interrupt requests5 more quickly. This may help reduce latency. Open
/etc/default/grub. Inside the quotation marks at the end of the line which starts with
threadirqs, and reboot.
Further latency reduction: run a low-latency or real-time kernel
If none of these help, you might try running a low-latency kernel. You can attempt to compile and use low-latency or real-time kernels; the Ubuntu Studio project provides them, and there are packages available for Debian. If you’ve come this far, though, I’d recommend the…
All-of-the-above solution: run an audio-focused Linux distribution
AV Linux is a Linux distribution focused on audio and video production. As such, it already employs the three tactics given above. It also includes a large amount of preinstalled, free, open-source AV software. It isn’t a daily driver distribution; rather, its foremost purpose is to be an audio or video workstation. It worked perfectly for me out of the box, and met my real-time monitoring and audio playback requirements for The Crossbox Podcast6. I recommend it wholeheartedly.
Given that my laptop is not primarily a podcast production device, I decided to carve a little 32gb partition out of the space at the end of my Windows partition, and installed AV Linux there. It records to my main Linux data partition instead of to the partition to which it is installed, and seems perfectly happy with this arrangement.
So am I. Anyway, thanks for reading. I hope it helped!
- Two identical microphones actually makes it slightly (though not insurmountably) harder, since they end up with the same ALSA name. ↩
- If you don’t have a package manager, you ought to be smart enough to know where to look. ↩
- This is most likely not because of your CPU, but rather because your Linux kernel does not have sufficient low-latency features to manage moving audio at the speeds we need it to move. ↩
- Remember, since we’re outputting to Loopback,1, that audio will be available for recording on Loopback,0. ↩
- Interrupt requests, or IRQs, are mechanisms by which hardware can interrupt a running program to run a special program known as an interrupt handler. Hardware sends interrupt requests to indicate that something has happened. Running them on independent threads improves the throughput, since more than one can happen at once, and, since they can be run on CPU cores not currently occupied, they interrupt other programs (like JACK) less frequently. ↩
- Expect us to hit our news countdown time cues a little more exactly, going forward. ↩
I was really excited when I ran across this post; it might solve all of my problems. But, I ran across an issue when running through the steps, not sure if you can help. When I run the alsa_in command for one mic, I get ‘selected sample format: 16 bit’. That is followed by, ‘delay = 14814’. About 3 minutes later, ‘delay = 8249’. And every 3 to 5 minutes, I get another delay message with a different number in the low, 8000’s. Any ideas what I am doing wrong and how to fix it?
There are a number of different causes of delays, but it’s my understanding that they generally crop up when something else is using the same interrupt request as the microphone in question. Wifi is a common culprit, so if you aren’t streaming your audio live, you might try disabling it. You may also want to try cranking the frame size up, or adding more frames per period. On my laptop’s Linux Mint partition, I have zero luck at 2048/2, but it works reasonably at 512/3, despite the latter being smaller buffers.
That said, since writing this post, I’ve found that non-audio-focused Linux distributions are hit and miss. Depending on just how the hardware is enumerated at boot time, my Mint partition might work perfectly, or might not work at all. If you’re running Ubuntu, there are official -preempt and -lowlatency kernels in the official repositories, and a -realtime kernel in a well-supported PPA.
Finally, for the podcast here, I switched over to a dedicated AVLinux install on a tiny partition, which records to my general Linux data partition. Ultimately, that’s the best solution in my limited experience, and I’m working on a second post on this topic which goes into more detail both on our setup here and how to replicate it.
As a final tangent, I’m not really an expert on this topic—just an amateur who’s had reason to look into it. For better advice, I would recommend trying the linuxmusicians.com forum. They know what they’re talking about.
I don’t recall offhand if this will send a notification, but the next article is finally finished if you should chance to see this. https://soapbox.manywords.press/2018/02/01/cheap-high-quality-podcast-recording/
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