After we got an overview about the peculiarities of the Linux system with respect to audio and recording in the last post we will dive some more into the Linux sound system itself. If you do not understand every last bit of information in here, don’t worry.
The Linux way of handling hardware
I don’t want to go into too much detail here but instead just hook you up with the general concepts of hardware handling in Linux. This will become quite handy whenever your sound card is not properly addressed by your system and you want to search for a solution online.
The Linux system (or the Linux kernel, to be more precise) needs two things to properly communicate with a sound card: a driver and a firmware. A driver is something like a protocol mapping a general interface to the card’s internal representation. Therefore drivers enable all cards to speak the same language. Sound card drivers are collected within the Advanced Linux Sound Architecture (ALSA) package, which itself is part of the Linux kernel. So they are already installed on your system and you are most probably already good to go. Sometimes you need to install some firmware, which is more or less a set of configuration to initialize the communication with the card, from some non-free repositories. But beware the blob!
After initializing the communication the Linux system manages the access of the sound card according to one of its core concepts:
Everything is a file.
All the entry points will be stored in /dev/snd/ while /proc/asound/cards holds a list of recognized cards.
But enough on the Linux system itself. To wrap things up, I compile a short list of bash commands you’ll find helpful for debugging and then let’s continue with the actual Linux sound servers.
- lshw - Compiles a list of all available hardware
- lspci -nn - Very detailed list of information about every connection established via PCI and PCIe
- lsmod - Lists all available kernel modules. The ones with snd in their names are commonly the ones responsible for the handling of the sound card
- modprobe - Loads an inactive kernel module
- ls /lib/firmware - Location of the firmware
- dmesg - Prints all system messages. Quite helpful in combination with removing and adding your USB sound card
Overall structure of the Linux sound servers and basic ALSA commands
While the ALSA package serves as a bridge to the hardware we will actually use two other systems for handling the sound: JACK and PulseAudio. Those sound servers will talk via ALSA with your sound card. So whenever you can’t access your sound card from within Ardour or Audacity it’s most probably the ALSA system itself and not the sound server on top having an issue. Formerly ALSA was intended to provide its own sound server too but its configuration is quite cumbersome and I would discourage you from using it. If you are still up for it, check out the basic introduction of concepts and acronyms on Wikipedia, the official and unofficial documentation, as well as ArchLinux’ take on personalizing ALSA’s ~/.asoundrc configuration file.
In addition ALSA provides a number of bash commands, which are quite handy for checking if your sound cards are working properly or not.
- alsamixer - A command line mixer for controlling volume and the selection of the sound card
- aplay –list-devices - Returns a list of all available audio devices
- *aplay -D CARDNAME AUDIOFILE.wav|.flac - Playback of a audio file via a specific sound card. It can be referenced with its full name or the its number (aplay –list-devices return card NUMBER: …) using hw:NUMBER. Note, you can only play uncompressed sound files using aplay! Mp3 won’t work.
The PulseAudio sound server
PulseAudio is the sound server already installed on most contemporary Linux systems per default. In contrast to ALSA it comes with built-in software mixing enabling you to play music from more than one audio source simultaneously. All those signals will be combined and sent to the sound card via the ALSA sink. It offers a bunch of modules handling the audio stream and is able to transmit audio signals via your local network.
In addition it offers some quite intuitive graphical user interfaces, like pavucontrol (volume control and sound card selection), paprefs (network access), and pasystray (status bar icon for quick control). Usually these programs already deal with all configurations you are interested in.
Alternatively you can interact with PulseAudio via the command line in the following way.
- pulseaudio -k or killall pulseaudio - Stopping the PulseAudio server. If ‘autospawn’ is activated, the sound server will restart as soon as any application on your computer tries to connect with it. You can override this behavior by changing your local copy (~/.config/pulse/client.conf) of the /etc/pulse/client.conf file.
- pulseaudio -D - Starts the server as a daemon (it will run in the background and will continue even if you close the command line)
- paplay - PulseAudio’s version of ALSA’s aplay
- pacmd - Will open an interactive shell to control the PulseAudio server
For a complete list of all pacmd commands run man pulse-cli-syntax in your bash.
PulseAudio itself is controlled via four different script located in /etc/pulse. If you intend to alter their content, please make a local copy in ~/.config/pulse and edit those files instead. The daemon.conf file sets the basic parameters of the sound server, like the sample rate, their format, and the handling of system resources, while the client.conf file controls the default behavior of how application will interact with the PulseAudio server. default.conf and system.conf are basically compilations of commands available through pacmd which will be executed on startup of the sound server. The ‘system’ way of startup is an alternative barely ever used (but if you have a static command line device, this one might be interesting for you).
The JACK sound server
Jack is the sound server of choice if you want to establish a more sophisticated workflow or intend to do some professional audio recording or production in Linux. It handles the access of the sound card via callbacks, what not just reduces the overall latency significantly but also ensures that all the different programs in your effect chain will be called in order and are synchronized. By connecting both your recording application and your drum sequencer as clients to the JACK servers, they will be perfectly synchronized and you will have a drum playback whenever you start your recording session without integrating it into the recording program itself.
One of the most convenient ways to configure JACK is to use QJackCtl or as an alternative Cadence and its siblings from the KXStudio suite. For a more detailed configuration please have a look into the official wiki or ArchLinux’s take.
A running JACK server does not necessarily return any audio. It, instead, serves as a convenience layer within your system all programs can both sent audio to and receive specific audio from. In other words: you can pipe the output of an arbitrary JACK-aware application into another one. To establish those connections (between programs as well as to the system’s sink and source), you can either use the connection manager of QJackCtl or an external one, like Patchage. Worth mentioning in the context of the graphical configuration and controlling of you JACK server are also JackEQ, JACK Rack, and the Non Session Manager.
When configuring JACK you will face a number terms, which, at least for me, are quite confusing in the beginning, like buffersize, bufferrate, frames, periods, latency, or samplerate. Let me shed some light on them and their connections.
Let’s illustrate them on an example. Imagine you are recording some audio, which enters your computer via a sound card in a particular way; microphone, line out, it doesn’t matter. In your computer you pipe the signal coming from the sound card into a chain of connected JACK client, and, finally, the signal will be sent to your sound card again in order to hear the results. The moment the audio signal enters your computer it has to be converted from its analog representation, a physical wave front or oscillation in the air pressure, into a digital one containing only zeros and ones. This will be done by sampling the signal using a specific samplerate (-r or –rate option). With a typical samplerate of 48000Hz the wave front will be measured 48000 times per second and each time its deflection will be stored on disk as a float number. Increasing the samplerate will therefore increase the number of floats used to approximate the analog signal and thus improves its digital representation. A single number written out during the measurement will be called a frame. Thus, with a samplerate of 48000Hz the audio device will get 48000 new frames of audio data per second.
Ideally, we would now pipe each frame through the chain of JACK clients and write the new numbers representing the output signal to the output port of our sound card. But this would be way too expensive in terms of CPU time and usage. Instead, we accumulate a number of frames in a buffer and provide it to the JACK clients so they can perform their operations on all frames in the buffer at once. This is way more efficient, but the signal will need more time to travel through the computer because of the accumulation process. So, we have to find a certain number of frames, the so-called buffersize or period (-p or –period option), that is a good trade-off between CPU usage and the waiting time. Unfortunately, there is no magical number working with all computers and/or sound cards. If your sound card supports hardware playback, you can set the buffersize to 1024, loose the realtime capabilities, but get a smooth signal instead. Otherwise, you have to gradually decrease the number in powers of two to find a setting that just does not produce audio glitches, so-called xruns, yet.
The amount of time the signal spends after its digitization until the end of JACK process chain is the input latency, which is calculated by the buffersize divided by the samplerate and given in seconds. The more frames we will combine into one buffer, the longer we have to wait for our signal. The more frames we sample per seconds, the faster the signal will reach the end of the chain for a fixed buffersize. But this is not the whole story yet. Since converting the digital signal back to an analog one does require some more time, not every buffer will be written out to the sound card. Instead, a number of buffers n (-n or –nperiods option) will be collected and all of them will then be handed to the sound card at once. With a usual setting of n=2 the output latency, which is calculated by the buffersize times n divided by the samplerate, will be twice as big as the input one. And since this is the time you spend waiting for your signal to reach the speakers, it’s most probably the latency you care most about.
The interplay between PulseAudio and JACK
Depending on your configuration at home you might find both the PulseAudio and JACK sound server attractive, each for a different scenario. Fortunately there are a number of options how those two can play together.
I am for example using my laptop with an external USB sound card for recording audio. I do not really like the idea of using a JACK setup running after system startup with my external sound card as default. The moment the card is not attached to my laptop anymore I have reconfigure the setup manually. Instead I can recommend to use the 4th option: to suspend PulseAudio when one starts JACK to record some audio.
In Setting > Server Prefix in QJackCtrl enter pasuspender – jack_control start. Instead of the classical JACK daemon jackd this approach is using D-Bus for a smoother transition between JACK and PulseAudio. To also have a smooth transition back to PulseAudio you can use the following script in Options > Scripting > Execute script after shutdown.
Hint: the configuration file for jackd is located in ~/.jackdrc and the corresponding one for jackdbus in ~/.config/jack/conf.xml.
Update (13.11.2018): Explaining the basic terms and parameters of the JACK server.