The Linux audio stack demystified
The article explores the Linux audio stack, detailing sound properties, digital audio sampling, and components like ALSA and PulseAudio, providing a comprehensive guide to audio processing in Linux systems.
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The article provides an in-depth exploration of the Linux audio stack, aiming to clarify the complexities of digital audio processing. It begins with a fundamental understanding of sound, describing it as vibrations that travel through mediums like air, and explains key properties such as period, amplitude, and frequency. The human auditory system's processing of sound waves is detailed, highlighting how sound is captured, amplified, and converted into electrical signals that the brain interprets.
The discussion then shifts to digital audio, emphasizing the importance of sampling, which converts analog sound into digital data. The Nyquist-Shannon Sampling Theorem is introduced, explaining the necessary sampling rates for accurate signal reconstruction. Quantization is also covered, illustrating how digital audio represents sound with a limited number of discrete values based on bit depth, with 16-bit and 24-bit audio being common standards.
The article further delves into the components of the Linux audio stack, including ALSA, JACK, PulseAudio, and PipeWire, each serving distinct roles in audio management. It explains the functions of sound servers, such as mixing multiple input streams and providing volume control. The piece concludes by encouraging readers to consider which sound server best suits their needs, emphasizing the interconnectedness of sound theory and practical audio processing in the Linux environment. Overall, the article serves as a comprehensive guide for those interested in understanding the intricacies of audio on Linux systems.
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The Linux audio stack demystified (and more)
Digital audio processing in Linux is complex, involving ALSA, JACK, PulseAudio, and PipeWire. Understanding requires knowledge of sound basics, human perception, and digital workings, including sampling and quantization.
- Readers found the article informative, especially for understanding the Linux audio stack and its components.
- There are requests for more detailed information on specific topics, such as the rating chart and Bluetooth audio issues.
- Some users express frustration with the complexity of audio processing on Linux compared to other systems.
- Concerns about real-time audio processing and the challenges of achieving low latency are highlighted.
- Several comments mention the need for clarity and simplicity in the audio stack, with nostalgia for older systems like OSS.
13 comments"At first Linus created /dev/dsp, and the user did smile upon him, and the user did see that it was good, and the user did see that it was simple, and people did use their sound, and people did pipe in and out sound as they did please, and Ken Thompson Shined upon them for following the way"
"Then the fiends got in on it and ruined it all, with needless complexities and configurations and situationships, with servers and daemons, and server and daemon wrappers to wrap the servers and daemons, and wrappers for those server wrappers, and then came security permissions for the server wrapper wrapper wrappers, why doesn't my sound work anymore, and then the server wrapper server wrapper wrapper server did need to be managed for massive added complexity, so initd was replaced by systemd, which solves the server wrapper wrapper server server wrapper through a highly complicated system of servers and services and wrappers"
RIP /dev/dsp you will be missed
- Kernighan 3:16
I’d like to see some more detail on the rating chart, particularly on the axes where pipewire doesn’t surpass JACK/pulseaudio.
As an embedded software engineer who deals with processing at hundreds of kilohertz, it is funny hearing anything running Linux called “real time”.
If it’s not carefully coded on bare metal for well understood hardware, it’s not real time, it’s just low latency. No true Scotsman though(looking over my shoulder for the FPGA programmers).
The problem with audio is it's realtime (isochronous), which means good audio processing requires a guarantee of sorts. To get that guarantee requires a path through the system that's clear, which can be difficult to construct.
Beats the pants off DANTE.
Related
Real-time audio programming 101: time waits for nothing
Real-time audio programming demands adherence to principles like avoiding blocking operations and prioritizing worst-case execution time to prevent glitches. Buffer sizes of 1-5ms are typical for low-latency software.
The Linux audio stack demystified (and more)
Digital audio processing in Linux is complex, involving ALSA, JACK, PulseAudio, and PipeWire. Understanding requires knowledge of sound basics, human perception, and digital workings, including sampling and quantization.