Introduction to Sound Programming with ALSA
ALSA stands for the Advanced Linux Sound Architecture. It consists of a set of kernel drivers, an application programming interface (API) library and utility programs for supporting sound under Linux. In this article, I present a brief overview of the ALSA Project and its software components. The focus is on programming the PCM interfaces of ALSA, including programming examples with which you can experiment.
You may want to explore ALSA simply because it is new, but it is not the only sound API available. ALSA is a good choice if you are performing low-level audio functions for maximum control and performance or want to make use of special features not supported by other sound APIs. If you already have written an audio application, you may want to add native support for the ALSA sound drivers. If your primary interest isn't audio and you simply want to play sound files, using one of the higher-level sound toolkits, such as SDL, OpenAL or those provided in desktop environments, may be a better choice. By using ALSA you are restricted to using systems running a Linux kernel with ALSA support.
The ALSA Project was started because the sound drivers in the Linux kernel (OSS/Free drivers) were not being maintained actively and were lagging behind the capabilities of new sound technology. Jaroslav Kysela, who previously had written a sound card driver, started the project. Over time, more developers joined, support for many sound cards was added and the structure of the API was refined.
During development of the 2.5 series of Linux kernel, ALSA was merged into the official kernel source. With the release of the 2.6 kernel, ALSA will be part of the stable Linux kernel and should be in wide use.
Sound, consisting of waves of varying air pressure, is converted to its electrical form by a transducer, such as a microphone. An analog-to-digital converter (ADC) converts the analog voltages into discrete values, called samples, at regular intervals in time, known as the sampling rate. By sending the samples to a digital-to-analog converter and an output transducer, such as a loudspeaker, the original sound can be reproduced.
The size of the samples, expressed in bits, is one factor that determines how accurately the sound is represented in digital form. The other major factor affecting sound quality is the sampling rate. The Nyquist Theorem states that the highest frequency that can be represented accurately is at most one-half the sampling rate.
ALSA consists of a series of kernel device drivers for many different sound cards, and it also provides an API library, libasound. Application developers are encouraged to program using the library API and not the kernel interface. The library provides a higher-level and more developer-friendly programming interface along with a logical naming of devices so that developers do not need to be aware of low-level details such as device files.
In contrast, OSS/Free drivers are programmed at the kernel system call level and require the developer to specify device filenames and perform many functions using ioctl calls. For backward compatibility, ALSA provides kernel modules that emulate the OSS/Free sound drivers, so most existing sound applications continue to run unchanged. An emulation wrapper library, libaoss, is available to emulate the OSS/Free API without kernel modules.
ALSA has a capability called plugins that allows extension to new devices, including virtual devices implemented entirely in software. ALSA provides a number of command-line utilities, including a mixer, sound file player and tools for controlling special features of specific sound cards.
The ALSA API can be broken down into the major interfaces it supports:
Control interface: a general-purpose facility for managing registers of sound cards and querying the available devices.
PCM interface: the interface for managing digital audio capture and playback. The rest of this article focuses on this interface, as it is the one most commonly used for digital audio applications.
Raw MIDI interface: supports MIDI (Musical Instrument Digital Interface), a standard for electronic musical instruments. This API provides access to a MIDI bus on a sound card. The raw interface works directly with the MIDI events, and the programmer is responsible for managing the protocol and timing.
Timer interface: provides access to timing hardware on sound cards used for synchronizing sound events.
Sequencer interface: a higher-level interface for MIDI programming and sound synthesis than the raw MIDI interface. It handles much of the MIDI protocol and timing.
Mixer interface: controls the devices on sound cards that route signals and control volume levels. It is built on top of the control interface.
|openHAB||Apr 24, 2017|
|Omesh Tickoo and Ravi Iyer's Making Sense of Sensors (Apress)||Apr 21, 2017|
|Low Power Wireless: 6LoWPAN, IEEE802.15.4 and the Raspberry Pi||Apr 20, 2017|
|CodeLathe's Tonido Personal Cloud||Apr 19, 2017|
|Wrapping Up the Mars Lander||Apr 18, 2017|
|MultiTaction's MT Canvus-Connect||Apr 17, 2017|
- Teradici's Cloud Access Platform: "Plug & Play" Cloud for the Enterprise
- Low Power Wireless: 6LoWPAN, IEEE802.15.4 and the Raspberry Pi
- The Weather Outside Is Frightful (Or Is It?)
- Simple Server Hardening
- Understanding Firewalld in Multi-Zone Configurations
- Gordon H. Williams' Making Things Smart (Maker Media, Inc.)
- Non-Linux FOSS: Control Web-Based Music!
- Server Technology's HDOT Alt-Phase Switched POPS PDU
- IGEL Universal Desktop Converter