Introduction to Sound Programming with ALSA

From Issue #126
October 2004

Sep 30, 2004 By Jeff Tranter

Audio/Video

Make maximum use of all the functionality in the new 2.6 kernel sound architecture using a simple API.

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.

History of ALSA

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.

Digital Audio Basics

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 Basics

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.

ALSA Architecture

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.

______________________

Comments

CROSS_COMPILE

Submitted by Anonymous (not verified) on Sun, 11/07/2010 - 06:30.

Hi,

I compile the source codes on my pc(ubuntu) with gcc play.c -o play -lasound and test with ./play < sample.raw. It works properly. But i want to use it on my beagleboarg(ARM platform). I tried to compile with
arm-none-linux-gcc play.c -o play -lasound
it doesn't work. It gives
/opt/arm-2009q1/bin/../lib/gcc/arm-none-linux-gnueabi/4.3.3/../../../../arm-none-linux-gnueabi/bin/ld: cannot find -lasound
collect2: ld returned 1 exit status

what is my mistake? Please help me.

Fatih

CROSS_COMPILE

Submitted by rangel (not verified) on Tue, 11/16/2010 - 13:34.

try: "strace ./play < sample.raw"
or "ldd play" and view if the correct path of library is linking

Volume needs to increase

Submitted by djguha (not verified) on Sat, 09/18/2010 - 09:53.

It is brilliant tutorial for beginners.I have run the capture and playback code.I noticed my volume level is very low when I capture my voice followed by playback code.But if I playback http://freewavesamples.com/files/Roland-JD-990-Windchimes.wav file, the sound volume is very good.I have followed a code to get the Capability of /dev/dsp from: http://www.oreilly.de/catalog/multilinux/excerpt/ch14-09.htm .
Here is the output :
Information on /dev/dsp:

Defaults:
sampling rate: 8000 Hz
channels: 1
sample size: 8 bits
block size: 4097 bytes

Supported Formats:
mu-law
unsigned 8-bit (default)
signed 16-bit little-endian
signed 16-bit big-endian
signed 8-bit
unsigned 16-bit little-endian
unsigned 16-bit big-endian

Capabilities:
revision: 1
full duplex: yes
real-time: yes
batch: no
coprocessor: no
trigger: yes
mmap: yes

Modes and Limits:
Device Sample Minimum Maximum
Channels Size Rate Rate
-------- -------- -------- --------
1 8 1000 100000
1 16 1000 100000
2 8 1000 100000
2 16 1000 100000

Please let me know how do I increase volume while capturing from /dev/dsp.
With regards,
-- Guha.

Example 3

Submitted by Milos (not verified) on Tue, 06/22/2010 - 06:09.

Hi to all.... Very good article for ALSA bigginers. I was trying to run this example on my PC. I want to play my .wav file (I will use file bugs_bye.wav as my input) and to hear that. When I run this example ./example3 bugs_bye.wav program executes but I can't hear anything. I am using file descriptor of this file insted of 0 (stdin) when I call function read like this:
int fd = open( argv[1] , O_RDONLY );
......
rc = read(fd, buffer, size);

The rest is same as example 3. What am I wrong??? Can someone help??? I am little bit confused.

Milos

Very Much informative

Submitted by Anonymous on Mon, 05/24/2010 - 07:19.

Thanks for this article

listing3

Submitted by Anonymous on Wed, 04/28/2010 - 10:11.

hi,i have a problem,i'm verry new to alsa programming,i didn't understand exactly how listing3 works.how do i put data into application buffer,to be transmitted to the sound card???i assume that the program is running correctly,but i don't know what to do after i run the program.Can anyone help me??

Re: Listing 3

Submitted by Anonymous on Thu, 04/29/2010 - 13:42.

Hello,

The program allocates a buffer,

buffer = (char *) malloc(size);

then in the loop it fills it from file descriptor 0,
i.e. stdin,

rc = read(0, buffer, size);

so to test, you could feed it something from the
command line.

It is set up for "CD" audio, 44100Hz, 2ch. I grabbed a sample here,

$ wget http://freewavesamples.com/files/Roland-JD-990-Windchimes.wav

I saved listing 3 as "listing3.c" and compiled,

$ gcc $(pkg-config --cflags --libs alsa) listing3.c -o listing3

Then to test,

$ cat Roland-JD-990-Windchimes.wav | ./listing3

If you wanted to generate your own sound, you could fill the buffer programmatically instead of reading from stdin.

Hope this helps.

[email protected]

Kudoos and hats off for the Article

Submitted by Anonymous on Wed, 02/17/2010 - 01:18.

I had been ssearching on net just to know how to start about with ALSA.

This article is simply great, starts from scratch building foundation and then takes you as far as you want to go..

Great work..

Regards
samir

question about one command of the program

Submitted by manu (not verified) on Thu, 07/30/2009 - 06:14.

hello,

I run your 4 programs coding in C with linux but i don't understand how does it work the next command :

snd_pcm_hw_params_get_period_time(params, &Val, &dir) who is used like that in Example 3 for example:

/* We want to loop for 5 seconds */
snd_pcm_hw_params_get_period_time(params,
&val, &dir);
/* 5 seconds in microseconds divided by
* period time */
loops = 5000000 / val;

If I add some lines of code, I can read the value of the variable "val" after and before the function " snd_pcm_hw_params_get_period_time(params, &val, &dir); ".

I can read that "val" before this function as a value 44099 (sample frequency minus 1), and after, "val" has the value 21333 at each time, whatever the others values enter as a parameters.

I don't understand how this function works and i would to know.

Would someone help me?

Thanks

Manu

5 sec

Submitted by MK (not verified) on Thu, 04/09/2009 - 10:06.

/* 5 seconds in microseconds divided by
* period time */
loops = 5000000 / val;

How did he calculate this?

how to read & play a sound file

Submitted by Shubham (not verified) on Thu, 02/05/2009 - 06:58.

hi,

I am new to ALSA programing. Could anyone tell how i can read & play a sound file using above example code for playback?

how to read & play a sound file

Submitted by Anonymous (not verified) on Thu, 07/15/2010 - 03:11.

You complie the example Listing 4 then plug the Headpone in the system.firsly check that your headphone is properly working by running the sound recorder program.after compiling code some output file is generated in the particular folder.
just start recording by typing ./.outfile name > sound.raw and record your voice through headphone for 5 sec.
For plaback compile code for playback in the seprate folder and run
./.outfile name

Converting to ALSA

Submitted by Anonymous (not verified) on Sat, 01/31/2009 - 02:19.

Firstly, this article is really good.

I need to write an application for FXS interface. the drivers are implements using ioctl system calls. How difficult is it to convert to ALSA API.
Or the otherway, what is to be done if i need to access these drivers in an application which is already supporting ALSA.

ok I just found it. for code

Submitted by Elie (not verified) on Mon, 12/29/2008 - 23:24.

ok I just found it. for code called 'alsa.c' it's

gcc alsa.c -lasound

Linking Libraries

Submitted by Elie (not verified) on Mon, 12/29/2008 - 22:53.

Umm... what libraries do i need to link? I tried the code (listings 3 & 4) and it compiled but didn't link. What gcc parameters need to be added?

(currently using Ubuntu Feisty 7.04)

undocumented bs wtf!

Submitted by mark manning (not verified) on Thu, 10/16/2008 - 12:22.

your second piece of example code has the following funciton call..

rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);

looking at the API reference for this call we see that the last parameter in this call is int mode. Mode may be one of the following values.

SND_PCM_NONBLOCK or SND_PCM_ASYNC. The first of these is equal to 0x01 and the second is equal to 0x02. WHAT is this undocumented mode zero that your example code AND the alsa example code uses.

undocumented == BAD MOJO!

great article tho :)

i am working on a project

Submitted by Anonymous (not verified) on Thu, 10/09/2008 - 21:38.

i am working on a project that needs a simple voice recording to be saved to a file, before further processing can be done on it... i am new to alsa (sound programming in general).
i believe alsa and oss can accomplish what i want but i need further guidance.
thanks.

ALSA Duplex Working code

Submitted by Santoshkumar (not verified) on Fri, 09/26/2008 - 06:24.

Does any body tested ALSA duplex (Record and playback)? Can you please point the place where i can get some reference source code?

Thank you,
Santosh
[email protected]

Updated code for FC6?

Submitted by William Estrada (not verified) on Fri, 09/14/2007 - 15:31.

I read this article and tried to run the sample code. It did compile and run but the record program produces garbage/noise. Is there an update for a Fedora Core 6 kernel? What kind of sound file does this code produce, wav, au, etc?

I don't know if anyone is

Submitted by Anonymous (not verified) on Wed, 07/25/2007 - 13:13.

I don't know if anyone is reading this.

But there are a few problems.

It records as far as I can see, but I get a lot of random garbage data which I don't want when recording. Anything I record is messed up in random data.

I´ve tried Sound

Submitted by David (not verified) on Wed, 12/20/2006 - 08:29.

I´ve tried Sound Programming with ALSA in a seminar, but i´m afraid, it is just too difficult for me. But i have to point out that i´m progging since one year, so i think i need just a bit more time. patience is all :-)

in listing 2, dir should be initialized

Submitted by Fabrice Pardo (not verified) on Wed, 12/13/2006 - 02:42.

Very useful article.

In listing 2, you should replace
int dir;
by
int dir = 0;
or replace &dir by NULL later

In my first test case, dir was randomly initialized to a negative
value, giving as a result
rate = 44099 bps

Actually, the snd_pcm_hw_params_set_rate_near function is poorly documented.

Display format

Submitted by MikeW (not verified) on Thu, 09/22/2005 - 10:56.

It would be nice if the (fixed) page width were a little greater, without having to shrink the browser text size.

Hard to read code that's full of line wraps.

Problem in opening default device in listing2 of this article

Submitted by Nagaraja S (not verified) on Fri, 03/11/2005 - 04:56.

Hi guys am new to this ALSA, I downloaded all driver, library utils from www.alsa-project.com and i installed.

when i run the listing1 of this doc it went fine and when i tried to run the second listing it says like this.
Unable to open the default device No such file.And i am using slackware with linux 2.4.22
Please help me out guys.

Thanx in advance for help

Re: Introduction to Sound Programming with ALSA

Submitted by JKCunningham (not verified) on Wed, 09/29/2004 - 01:00.

Great article. I tried the first source and it worked fine, but the second bombed:

> ./listing_2
unable to set hw parameters: Invalid argument

Any idea why this would be? I didn't modify the source.

-Jeff

Problem in listing2 program Unable to open default device

Submitted by Nagaraja (not verified) on Fri, 03/11/2005 - 04:18.

Hi guys am new to this ALSA, I downloaded all driver, library utils from www.alsa-project.com and i installed.

Thanx in advance for help

Re: Introduction to Sound Programming with ALSA

Submitted by Pablo (not verified) on Sat, 11/27/2004 - 17:16.

It happened the same to me. When I change the sampling frequency value to 48 kHz then I got the error message:
unable to set hw parameters: Invalid argument
For any other sampling frequency it works ok.

Pablo

It happened the same to me. W

Submitted by Pablo (not verified) on Sat, 11/27/2004 - 17:15.

Pablo

Re: Introduction to Sound Programming with ALSA

Submitted by Anonymous on Fri, 10/01/2004 - 01:00.

Someone has posted a note at the Resources page
,
saying that the variable "dir" needs to be initialized to 0 (zero).
That seemed to solve my problems, at least.

Re: Introduction to Sound Programming with ALSA

Submitted by Anonymous on Fri, 10/01/2004 - 01:00.

I am having the same problems with listing 4. (And it seems that others have too, see the Resources page.) In my case the problem seems to be the sampling frequency, if I change 44100 til 88200 the program will run.

What I really would like to know is whether this is a problem with the example program from the article (I did not modify it), a problem with ALSA (I use Debian Sarge with 2.6-kernel) or a problem with my sound card (SoundBlaster Live).

problem

Submitted by Anonymous (not verified) on Sun, 11/07/2010 - 13:30.

Hi,

I tried to run the codes under beagleboard. I got the error message:
unable to set hw parameters: Invalid argument
I change the sampling frequency from 44100 to 88200 even to 132300 as you said. This time it works but when i run
./example3 < /dev/urandom
this command it gives me veri bad sound like "CCcczzZZZZzzzrrttTtTT".

My beagleboards parameters are this : //my pc(ubuntu)

PCM handle name = 'default'
PCM state = PREPARED
access type = RW_INTERLEAVED
format = 'S16_LE' (Signed 16 bit Little Endian)
subformat = 'STD' (Standard)
channels = 2
rate = 44100 bps
period time = 5804 us //23219
period size = 256 frames //1024
buffer time = 743038 us //23219
buffer size = 32768 frames //1048576
periods per buffer = 128 frames //1024
exact rate = 44100/1 bps
significant bits = 16
tick time = 16 us
is batch = 0
is block transfer = 0 //1
is double = 0
is half duplex = 0
is joint duplex = 0
can overrange = 0
can mmap = 1 //0
can pause = 1
can resume = 1 //0
can sync start = 0

The parameters which are bold are difference between my pc(ubuntu).
Please Can you give me a suggestion how to modify the code?

Thx in Advance.

Fatih

problem

Submitted by Anonymous (not verified) on Sun, 11/07/2010 - 13:29.

Hi,

My beagleboards parameters are this : //my pc(ubuntu)

The parameters which are bold are difference between my pc(ubuntu).
Please Can you give me a suggestion how to modify the code?

Thx in Advance.

Fatih

Re: Introduction to Sound Programming with ALSA

Submitted by Anonymous on Fri, 09/24/2004 - 01:00.

A simple question: Why can't the ALSA developers make sure that ALSA provides sound support at least as good as OSS? After having tested it on a number of boards, it was always a pain in the neck to configure properly, and getting it to work, an adventure. I will stay with OSS until ALSA becomes much more user-friendly.

Re: Introduction to Sound Programming with ALSA

Submitted by Anonymous on Wed, 09/22/2004 - 01:00.

Excellent informatiom. Now lets have a simple real time mixer with chnangeable effect to really show what ALSA + 2.6 Linux can do

Re: Introduction to Sound Programming with ALSA

Submitted by Anonymous on Wed, 09/15/2004 - 01:00.

Yes, I agree, excellent article and a brilliant introduction to ALSA.

Keep up the good work!

Re: Introduction to Sound Programming with ALSA

Submitted by Anonymous on Thu, 09/09/2004 - 01:00.

Informative article. More intros to popular or obscure libraries please!

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