Introduction to Sound Programming with ALSA

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http://www.linuxjournal.com/article/6735
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.

Device Naming
The library API works with logical device names rather than device
files. The device names can be real hardware devices or plugins.
Hardware devices use the format
hw:i,j, where
i is the card number
and j
is the device on that card. The first sound device is
hw:0,0. The alias default refers to the first sound device and is
used in all of the examples in this article. Plugins use other unique
names; plughw:, for example, is a plugin that provides access to the
hardware device but provides features, such as sampling rate
conversion, in software for hardware that does not directly support it.
The dmix
and dshare plugins allow you to downmix several streams and split a
single stream dynamically among different applications.
Sound Buffers and Data Transfer
A sound card has a hardware buffer that stores recorded samples.
When the buffer is sufficiently full, it generates an interrupt. The
kernel sound driver then uses direct memory access (DMA) to transfer
samples to an application buffer in memory. Similarly, for playback, another application buffer is transferred from memory to
the sound card's hardware buffer using DMA.
These hardware buffers are ring buffers, meaning the data wraps
back to the start when the end of the buffer is reached. A
pointer is maintained to keep track of the current positions in both the
hardware buffer and the application buffer. Outside of the kernel,
only the application buffer is of interest, so from here on we
discuss only the application buffer.
The size of the buffer can be programmed by ALSA library calls. The
buffer can be quite large, and transferring it in one operation could result in unacceptable delays, called latency. To solve
this, ALSA splits the buffer up into a series of periods (called
fragments in OSS/Free) and transfers the data in units of a period.
A period stores frames, each of which contains the samples captured at
one point in time. For a stereo device, the frame would contain
samples for two channels. Figure 1 illustrates the breakdown of a
buffer into periods, frames and samples with some hypothetical
values.
Here, left and right channel information is stored alternately within a
frame; this is called interleaved mode. A non-interleaved mode, where
all the sample data for one channel is stored followed by the data for
the next channel, also is supported.

Figure 1. The Application Buffer
Over and Under Run
When a sound device is active, data is transferred
continuously between the hardware and application buffers. In the case of data
capture (recording), if the application does not read the data in the
buffer rapidly enough, the circular buffer is overwritten with
new data. The resulting data loss is known as overrun. During playback, if
the application does not pass data into the buffer quickly enough, it
becomes starved for data, resulting in an error called underrun.
The ALSA documentation sometimes refers to both of these conditions
using the term XRUN. Properly designed applications can minimize
XRUN and recover if it occurs.
A Typical Sound Application
Programs that use the PCM interface generally follow this
pseudo-code:
open interface for capture or playback
set hardware parameters
(access mode, data format, channels, rate, etc.)
while there is data to be processed:
   read PCM data (capture)
   or write PCM data (playback)
close interface
We look at some working code in the following sections. I
recommend you compile and run these on your Linux system, look at the
output and try some of the suggested modifications. The full listings
for the example programs that accompany this article are available for
download from
ftp.ssc.com/pub/lj/listings/issue126/6735.tgz
.

Listing 1. Display Some PCM Types and Formats
#include
int main() {
  int val;
  printf("ALSA library version: %s\n",
          SND_LIB_VERSION_STR);
  printf("\nPCM stream types:\n");
  for (val = 0; val
Listing 1 displays some of the PCM data types and parameters used
by ALSA.
The first requirement is to include the header file that brings in the
definitions for all of the ALSA library functions. One of the
definitions is the version of ALSA, which is displayed.
The remainder of the program iterates through a number of PCM data
types, starting with the stream types. ALSA provides
symbolic names for the last enumerated value and a utility function
that returns a descriptive string for a value. As you can see in the
output, ALSA supports many different data formats, 38 for the version
of ALSA on my system.
The program must be linked with the ALSA library,
libasound, to run. Typically, you would add the option -lasound on the
linker command line. Some ALSA library functions use the dlopen
function and floating-point operations, so you also may need to add
-ldl and -lm.

Listing 2. Opening PCM Device and Setting Parameters
/*
This example opens the default PCM device, sets
some parameters, and then displays the value
of most of the hardware parameters. It does not
perform any sound playback or recording.
*/
/* Use the newer ALSA API */
#define ALSA_PCM_NEW_HW_PARAMS_API
/* All of the ALSA library API is defined
* in this header */
#include
int main() {
  int rc;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val, val2;
  int dir;
  snd_pcm_uframes_t frames;
  /* Open PCM device for playback. */
  rc = snd_pcm_open(&handle, "default",
                    SND_PCM_STREAM_PLAYBACK, 0);
  if (rc
Listing 2 opens the default PCM device, sets some parameters and
then displays the values of most of the hardware parameters. It does
not perform any sound playback or recording.
The call to snd_pcm_open opens the default PCM device and sets the
access mode to PLAYBACK. This function returns a handle in the
first function argument that is used in subsequent calls to
manipulate the PCM stream. Like most ALSA library calls, the function
returns an integer return status, a negative value indicating an error
condition. In this case, we check the return code; if it indicates
failure, we display the error message using the snd_strerror function
and exit. In the interest of clarity, I have omitted most of the error
checking from the example programs. In a production application, one
should check the return code of every API call and provide appropriate
error handling.
In order to set the hardware parameters for the stream, we need to
allocate a variable of type snd_pcm_hw_params_t. We do this with the
macro snd_pcm_hw_params_alloca. Next, we initialize the variable using
the function snd_pcm_hw_params_any, passing the previously opened
PCM stream.
We now set the desired hardware parameters using API calls that take
the PCM stream handle, the hardware parameters structure and the
parameter value. We set the stream to interleaved mode, 16-bit sample
size, 2 channels and a 44,100 bps sampling rate. In the case of
the sampling rate, sound hardware is not always able to support every
sampling rate exactly. We use the function
snd_pcm_hw_params_set_rate_near to request the nearest supported
sampling rate to the requested value. The hardware parameters are not
actually made active until we call the function snd_pcm_hw_params.
The rest of the program obtains and displays a number of the PCM
stream parameters, including the period and buffer sizes. The results
displayed vary somewhat depending on the sound hardware.
After running the program on your system, experiment and make
some changes. Change the device name from default to hw:0,0 or
plughw: and see whether the results change. Change the hardware parameter
values and observe how the displayed results change.

Listing 3. Simple Sound Playback
/*
This example reads standard from input and writes
to the default PCM device for 5 seconds of data.
*/
/* Use the newer ALSA API */
#define ALSA_PCM_NEW_HW_PARAMS_API
#include
int main() {
  long loops;
  int rc;
  int size;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  int dir;
  snd_pcm_uframes_t frames;
  char *buffer;
  /* Open PCM device for playback. */
  rc = snd_pcm_open(&handle, "default",
                    SND_PCM_STREAM_PLAYBACK, 0);
  if (rc  0) {
    loops--;
    rc = read(0, buffer, size);
    if (rc == 0) {
      fprintf(stderr, "end of file on input\n");
      break;
    } else if (rc != size) {
      fprintf(stderr,
              "short read: read %d bytes\n", rc);
    }
    rc = snd_pcm_writei(handle, buffer, frames);
    if (rc == -EPIPE) {
      /* EPIPE means underrun */
      fprintf(stderr, "underrun occurred\n");
      snd_pcm_prepare(handle);
    } else if (rc
Listing 3 extends the previous example by writing sound samples to the
sound card to produce playback. In this case we read bytes from
standard input, enough for one period, and write them to the sound
card until five seconds of data has been transferred.
The beginning of the program is the same as in the previous example—the
PCM device is opened and the hardware parameters are set. We use the
period size chosen by ALSA and make this the size of our
buffer for storing samples. We then find out that period time so
we can calculate how many periods the program should process in order
to run for five seconds.
In the loop that manages data, we read from standard input and fill
our buffer with one period of samples. We check for and handle errors
resulting from reaching the end of
file or reading a different number of bytes from what was expected.
To send data to the PCM device, we use the snd_pcm_writei call. It
operates much like the kernel write system call, except that the size
is specified in frames. We check the return code for a number of error
conditions. A return code of EPIPE indicates that underrun
occurred, which causes the PCM stream to go into the XRUN state and
stop processing data. The standard method to recover from this state is to
use the snd_pcm_prepare function call to put the stream in the
PREPARED state so it can start again the next time we write data to
the stream. If we receive a different error result, we display the
error code and continue. Finally, if the number of frames written is
not what was expected, we display an error message.
The program loops until five seconds' worth of frames has been
transferred or end of file read occurs on the input. We then call
snd_pcm_drain to allow any pending sound samples to be transferred, then
close the stream. We free the dynamically allocated buffer and exit.
We should see that the program is not useful unless the input is redirected to
something other than a console. Try running it with the device
/dev/urandom, which produces random data, like this:
./example3
The random data should produce white noise for five seconds.
Next, try redirecting the input to /dev/null or /dev/zero and compare the
results. Change some parameters, such as the sampling rate and data format, and
see how it affects the results.

Listing 4. Simple Sound Recording
/*
This example reads from the default PCM device
and writes to standard output for 5 seconds of data.
*/
/* Use the newer ALSA API */
#define ALSA_PCM_NEW_HW_PARAMS_API
#include
int main() {
  long loops;
  int rc;
  int size;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  int dir;
  snd_pcm_uframes_t frames;
  char *buffer;
  /* Open PCM device for recording (capture). */
  rc = snd_pcm_open(&handle, "default",
                    SND_PCM_STREAM_CAPTURE, 0);
  if (rc  0) {
    loops--;
    rc = snd_pcm_readi(handle, buffer, frames);
    if (rc == -EPIPE) {
      /* EPIPE means overrun */
      fprintf(stderr, "overrun occurred\n");
      snd_pcm_prepare(handle);
    } else if (rc
Listing 4 is much like Listing 3, except that we perform PCM
capture (recording). When we open the PCM stream, we specify the mode
as SND_PCM_STREAM_CAPTURE.
In the main processing loop, we read the samples from the sound
hardware using snd_pcm_readi and write it to standard output using
write. We check for overrun and handle it in the same manner as we did
underrun in Listing 3.
Running Listing 4 records approximately five seconds of data and
sends it to standard out; you should redirect it to a file.
If you have a microphone connected to your sound card, use a mixer
program to set the recording source and level. Alternatively, you can
run a CD player program and set the recording source to CD. Try running
Listing 4 and redirecting the output to a file. You then can run Listing
3 to play back the data:
./listing4 > sound.raw
./listing3
If your sound card supports full duplex sound, you should be able to pipe the
programs together and hear the recorded sound coming out of the sound
card by typing: ./listing4 | ./listing3.
By changing the PCM parameters you can experiment with the effect of
sampling rates and formats.
Advanced Features
In the previous examples, the PCM streams were operating in blocking
mode, that is, the calls would not return until the data had been
transferred. In an interactive event-driven application, this situation could
lock up the application for unacceptably long periods of time. ALSA
allows opening a stream in nonblocking mode where the read and write
functions return immediately. If data transfers are pending and the
calls cannot be processed, ALSA returns an error code of EBUSY.
Many graphical applications use callbacks to handle events. ALSA
supports opening a PCM stream in asynchronous mode. This allows
registering a callback function to be called when a period of sample
data has been transferred.
The snd_pcm_readi and snd_pcm_writei calls used here are similar to the
Linux read and write system calls. The letter i indicates that the
frames are interleaved; corresponding functions exist for
non-interleaved mode. Many devices under Linux also support the mmap
system call, which maps them into memory where they can be manipulated
with pointers. Finally, ALSA supports opening a PCM channel in mmap
mode, which allows efficient zero copy access to sound data.
Conclusion
I hope this article has motivated you to try some ALSA programming.
As the 2.6 kernel becomes commonly used by Linux distributions, ALSA
should become more widely used, and its advanced features should help
Linux audio applications move forward.
My thanks to Jaroslav Kysela and Takashi Iwai for reviewing a draft
of this article and providing me with useful comments.
Resources for this article:
/article/7705
.
Jeff Tranter has been using, writing about and contributing to Linux
since 1992. He works for Xandros Corporation in Ottawa, Canada.
               
               
               

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