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#
# Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
include ../Makefile.conf
TARGETS=soundapi.txt \
soundapi.html
all: $(TARGETS)
soundapi.txt: soundapi.sgml
sgml2txt soundapi.sgml
soundapi.html: soundapi.sgml
sgml2html soundapi.sgml
clean:
rm -f core .depend *.orig *~

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ALSA library plan
=================
=====
ST: Status of coding
NT: Note
RP: Responsible person
This file is maintained by Jaroslav Kysela <perex@jcu.cz>.
=====
Library code
============
1: Control interface routines
ST: Low-Level code done
RP: Jaroslav Kysela <perex@jcu.cz>
1.1: Network layer
2: MIXER routines
ST: Low-Level code done
RP: Jaroslav Kysela <perex@jcu.cz>
2.1: Network layer
3: PCM routines
ST: Low-Level code done
RP: Jaroslav Kysela <perex@jcu.cz>
3.1: Network layer + mixing routines to allow playback more than one input
to one exclusive device
3.2: Player routines for digital data (.wav,.snd,.mp3 etc...) (???)
4: Instrument loading routines
4.1: Network layer
4.2: Instrument format interface
4.2.1: GF1 patches (.pat)
4.2.2: AMD FFFF
4.2.3: SoundFont (for SoundBlaster)
4.2.4: FM/OPL3 instrument formats
5: Raw FM/OPL3 interface
6: SYNTH interface
6.1: Network layer
6.2: player for .MOD,.S3M etc. (???)
7: Instrument server interface
8: Instrument configuration routines for MIDI
9: raw MIDI interface
9.1: Network layer
10: sequenced MIDI interface
10.1: Network layer
10.2: player for MIDI files

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<HTML>
<HEAD>
<TITLE>Advanced Linux Sound Architecture - Library API: Introduction</TITLE>
</HEAD>
<BODY>
Previous
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<HR>
<H2><A NAME="s1">1. Introduction</A></H2>
<P>The Advanced Linux Sound Architecture comes with a kernel API & library API.
This document describes the library API and how it interfaces with the kernel
API. The kernal API will probably never be documented in standalone form.</P>
<P>Application programmers should use the library API rather than kernel API.
The Library offers 100% of the functionally of the kernel API, but add next
major improvements in usability, making the application code simpler and
better looking. In addition, some of the some fixes/compatibility code in,
may be placed in the library code instead of the kernel driver.</P>
<P>For a complete list of all variables and functions in the API you should look
at the following header files:
<OL>
<LI>/usr/include/sys/soundlib.h</LI>
<LI>/usr/include/linux/sound.h</LI>
<LI>/usr/include/linux/sounddetect.h</LI>
</OL>
</P>
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<HEAD>
<TITLE>Advanced Linux Sound Architecture - Library API: Error Codes</TITLE>
</HEAD>
<BODY>
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<HR>
<H2><A NAME="s2">2. Error Codes</A></H2>
<P>All functions return int (or some sort of signed value). If this value
is negative it represents an error code. Codes up to SND_ERROR_BEGIN (500000)
represents standard system errors. Codes equal or greather than this value
represents sound library API errors. All error codes begin with the prefix
<I>SND_ERROR_</I>.</P>
<H2><A NAME="ss2.1">2.1 Error Codes in Detail</A></H2>
<P>
<DL>
<DT><B>SND_ERROR_UNCOMPATIBLE_VERSION (500000)</B><DD><P>This error is caused if the driver uses an incompatible kernel API for this
interface and hence the library doesn't know how this API can be used.</P>
</DL>
</P>
<H2><A NAME="ss2.2">2.2 Functions</A></H2>
<H3>const char *snd_strerror( int errnum ) </H3>
<P>This functions converts error code to a string. Its functionality is the same
as the <I>strerror</I> function from the standard C library, but this
function returns correct strings for sound error codes, too.</P>
<HR>
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<HTML>
<HEAD>
<TITLE>Advanced Linux Sound Architecture - Library API: Control Interface</TITLE>
</HEAD>
<BODY>
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<HR>
<H2><A NAME="s3">3. Control Interface</A></H2>
<P>The control interfaces gives application various information about the
currently installed sound driver in the system. The interface should be used
to detect if another sound interface is present for selected soundcard or,
for example, to create a list of devices (MIXER, PCM etc) from which
the user can select.</P>
<H2><A NAME="ss3.1">3.1 Low-Level Layer</A></H2>
<H3>int snd_cards( void ) </H3>
<P>Returns the number of soundcards present in the system, if any. Otherwise
it returns a negative value, which maps to an error code. This function
will return 0 if no soundcards are detected.</P>
<H3>unsigned int snd_cards_mask( void ) </H3>
<P>Returns the bitmap of soundcards present in the system, if any. Otherwise
it returns a negative value, which maps to an error code. This function
will return 0 if no soundcards are detected. First soundcard is represented
with bit 0.</P>
<H3>int snd_ctl_open( void **handle, int card ) </H3>
<P>Creates a new handle and opens communication with the kernel sound
control interface for soundcard number <I>card</I> (0-N). The function
also checks if protocol is compatible, so as to prevent the use of old
programs with a new kernel API. Function returns zero if successful,
otherwise an error code is returned.</P>
<H3>int snd_ctl_close( void *handle ) </H3>
<P>Function frees all resources allocated with control handle and
closes the kernel sound control interface. Function returns zero if
successful, otherwise it returns an error code.</P>
<H3>int snd_ctl_file_descriptor( void *handle ) </H3>
<P>Function returns file descriptor for the kernel sound control interface.
This function should be used in very special cases. Function returns
a negative error code if some error was encountered.</P>
<H3>int snd_ctl_hw_info( void *handle, struct snd_ctl_hw_info *info ) </H3>
<P>Fills the info structure with data about the sound hardware referenced
by handle. Function returns zero if successful, otherwise it returns
an error code.
<HR>
<PRE>
#define SND_CTL_GCAPS_MIDI 0x0000001 /* driver has MIDI interface */
#define SND_CTL_LCAPS_SYNTH 0x0000001 /* soundcard has synthesizer */
#define SND_CTL_LCAPS_RAWFM 0x0000002 /* soundcard has RAW FM/OPL3 */
struct snd_ctl_hw_info {
unsigned int type; /* type of card - see SND_CARD_TYPE_XXXX */
unsigned int gcaps; /* see SND_CTL_GCAPS_XXXX */
unsigned int lcaps; /* see SND_CTL_LCAPS_XXXX */
unsigned int pcmdevs; /* count of PCM devices (0 to N) */
unsigned int mixerdevs; /* count of MIXER devices (0 to N) */
unsigned int mididevs; /* count of raw MIDI devices (0 to N) */
char id[8]; /* ID of card (user selectable) */
char name[80]; /* name/info text about soundcard */
unsigned char reserved[128]; /* reserved for future use */
};
</PRE>
<HR>
</P>
<H3>int snd_ctl_pcm_info( void *handle, int dev, snd_pcm_info_t *info ) </H3>
<P>Fills the *info structure with data about the PCM device. Function returns
zero if successful, otherwise it returns an error code. Details about
the snd_pcm_info_t structure are in the <B>Digital Audio (PCM) Interface</B>
section. The argument <I>dev</I> selects the device number for the
soundcard referenced by *handle. Its range is 0 to N where N is
<I>struct snd_ctl_hw_info -> pcmdevs - 1</I>. This function will work if
the selected PCM device is busy, too. It should be used to collect
information about PCM devices without exclusive lock.</P>
<H3>int snd_ctl_pcm_playback_info( void *handle, int dev, snd_pcm_playback_info_t *info ) </H3>
<P>Fills the *info structure with data about the PCM device and playback direction.
Function returns zero if successful, otherwise it returns an error code.
Details about the snd_pcm_playback_info_t structure are in the
<B>Digital Audio (PCM) Interface</B> section. The argument <I>dev</I>
selects the device number for the soundcard referenced by *handle. Its
range is 0 to N where N is <I>struct snd_ctl_hw_info -> pcmdevs - 1</I>.
This function will work if the selected PCM device is busy, too. It should
be used to collect information about PCM devices without exclusive lock.</P>
<H3>int snd_ctl_pcm_record_info( void *handle, int dev, snd_pcm_record_info_t *info ) </H3>
<P>Fills the *info structure with data about the PCM device and record direction.
Function returns zero if successful, otherwise it returns an error code.
Details about the snd_pcm_record_info_t structure are in the
<B>Digital Audio (PCM) Interface</B> section. The argument <I>dev</I>
selects the device number for the soundcard referenced by *handle. Its
range is 0 to N where N is <I>struct snd_ctl_hw_info -> pcmdevs - 1</I>.
This function will work if the selected PCM device is busy, too. It should
be used to collect information about PCM devices without exclusive lock.</P>
<H3>int snd_ctl_mixer_info( void *handle, int dev, snd_mixer_info_t *info ) </H3>
<P>Fills the *info structure with data about the mixer device. Returns zero
if successful, otherwise it returns an error code. Details about the
snd_mixer_info_t structure are in the <B>Mixer Interface</B> section.
The argument <I>dev</I> specifies the device number for the appropriate
soundcard. Its range is 0 to N where N found from
<I>struct snd_ctl_hw_info -> mixerdevs - 1</I>.
It should be used to collect information about mixer devices.</P>
<H2><A NAME="ss3.2">3.2 Examples</A></H2>
<P>The following example shows how all PCM devices can be detected for the first
soundcard (#0) in the system.</P>
<P>
<BLOCKQUOTE><CODE>
<HR>
<PRE>
int card = 0, err;
void *handle;
stuct snd_ctl_hw_info info;
if ( (err = snd_ctl_open( &amp;handle, card )) &lt; 0 ) {
fprintf( stderr, &quot;open failed: %s\n&quot;, snd_strerror( err ) );
return;
}
if ( (err = snd_ctl_hw_info( handle, &amp;info )) &lt; 0 ) {
fprintf( stderr, &quot;hw info failed: %s\n&quot;, snd_strerror( err ) );
snd_ctl_close( handle );
return;
}
printf( &quot;Installed PCM devices for card #i: %i\n&quot;, card + 1, info.pcmdevs );
snd_ctl_close( handle );
</PRE>
<HR>
</CODE></BLOCKQUOTE>
</P>
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<HTML>
<HEAD>
<TITLE>Advanced Linux Sound Architecture - Library API: Mixer Interface</TITLE>
</HEAD>
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<HR>
<H2><A NAME="s4">4. Mixer Interface</A></H2>
<P>The Mixer Interface allows applications to change the volume level of
a soundcard's input/output channels in both the linear range (0-100)
and in decibels. It also supports features like hardware mute, input
sound source, etc.</P>
<H2><A NAME="ss4.1">4.1 Low-Level Layer</A></H2>
<P>Mixer devices aren't opened exclusively. This allows applications to
open a device multiple times with one or more processes.</P>
<H3>int snd_mixer_open( void **handle, int card, int device ) </H3>
<P>Creates new handle and opens a connection to the kernel sound
mixer interface for soundcard number <I>card</I> (0-N) and mixer
device number <I>device</I>. Also checks if protocol is
compatible to prevent use of old programs with new kernel API. Function
returns zero if successful, otherwise it returns an error code.</P>
<H3>int snd_mixer_close( void *handle ) </H3>
<P>Frees all resources allocated to the mixer handle and
closes its connection to the kernel sound mixer interface. Function
returns zero if successful, otherwise it returns an error code.</P>
<H3>int snd_mixer_file_descriptor( void *handle ) </H3>
<P>Returns the file descriptor for the connection to the kernel sound
mixer interface. This function should be used only in very
special cases. Function returns a negative error code if an
error was encountered. </P>
<P>The file descriptor should be used for the <I>select</I> synchronous
multiplexer function for deterimeing read direction. Applications should
call <I>snd_mixer_read</I> function if some data is waiting to be read.
It is recomended that you do this, since it leaves place for this function
to handle some new kernel API specifications.</P>
<H3>int snd_mixer_channels( void *handle ) </H3>
<P>Returns the count of mixer channels for appropriate mixer device, otherwise
the return value is negative, and signifies an error code. Never returns
zero.</P>
<H3>int snd_mixer_info( void *handle, snd_mixer_info_t *info ) </H3>
<P>Fills the *info structure with information about the mixer associated with
*handle. Returns zero if successful, otherwise it returns an error code.
<HR>
<PRE>
#define SND_MIXER_INFO_CAP_EXCL_RECORD 0x00000001
struct snd_mixer_info {
unsigned int type; /* type of soundcard - SND_CARD_TYPE_XXXX */
unsigned int channels; /* count of mixer devices */
unsigned int caps; /* some flags about this device (SND_MIXER_INFO_CAP_XXXX) */
unsigned char id[32]; /* ID of this mixer */
unsigned char name[80]; /* name of this device */
char reserved[ 32 ]; /* reserved for future use */
};
</PRE>
<HR>
</P>
<H3>int snd_mixer_channel( void *handle, const char *channel_id ) </H3>
<P>Returns the channel number (index) associated with channel_id (channel name),
or returns an error code.
<HR>
<PRE>
#define SND_MIXER_ID_MASTER &quot;Master&quot;
#define SND_MIXER_ID_BASS &quot;Bass&quot;
#define SND_MIXER_ID_TREBLE &quot;Treble&quot;
#define SND_MIXER_ID_SYNTHESIZER &quot;Synth&quot;
#define SND_MIXER_ID_SYNTHESIZER1 &quot;Synth 1&quot;
#define SND_MIXER_ID_FM &quot;FM&quot;
#define SND_MIXER_ID_EFFECT &quot;Effect&quot;
#define SND_MIXER_ID_PCM &quot;PCM&quot;
#define SND_MIXER_ID_PCM1 &quot;PCM 1&quot;
#define SND_MIXER_ID_LINE &quot;Line-In&quot;
#define SND_MIXER_ID_MIC &quot;MIC&quot;
#define SND_MIXER_ID_CD &quot;CD&quot;
#define SND_MIXER_ID_GAIN &quot;Record-Gain&quot;
#define SND_MIXER_ID_IGAIN &quot;In-Gain&quot;
#define SND_MIXER_ID_OGAIN &quot;Out-Gain&quot;
#define SND_MIXER_ID_LOOPBACK &quot;Loopback&quot;
#define SND_MIXER_ID_SPEAKER &quot;PC Speaker&quot;
#define SND_MIXER_ID_AUXA &quot;Aux A&quot;
#define SND_MIXER_ID_AUXB &quot;Aux B&quot;
#define SND_MIXER_ID_AUXC &quot;Aux C&quot;
</PRE>
<HR>
</P>
<H3>int snd_mixer_exact_mode( void *handle, int enable ) </H3>
<P>Turns on or off (by default) exact mode. This mode allows to application
set/get volume values in exact range which uses hardware. In non-exact
mode is range always from 0 to 100 and conversion to hardware range does
driver. Function returns zero if successful, otherwise it returns an error
code.</P>
<H3>int snd_mixer_channel_info( void *handle, int channel, snd_mixer_channel_info_t *info ) </H3>
<P>Fills the *info structure. The argument <I>channel</I> specifies channel
(0 to N) for which is the info requested. Function returns zero if
successful, otherwise it returns an error code.
<HR>
<PRE>
#define SND_MIXER_CINFO_CAP_RECORD 0x00000001
#define SND_MIXER_CINFO_CAP_STEREO 0x00000002
#define SND_MIXER_CINFO_CAP_MUTE 0x00000004
#define SND_MIXER_CINFO_CAP_HWMUTE 0x00000008 /* channel supports hardware mute */
#define SND_MIXER_CINFO_CAP_DIGITAL 0x00000010 /* channel does digital (not analog) mixing */
#define SND_MIXER_CINOF_CAP_INPUT 0x00000020 /* input channel */
struct snd_mixer_channel_info {
unsigned int channel; /* channel # (filled by application) */
unsigned int parent; /* parent channel # or SND_MIXER_PARENT */
unsigned char name[12]; /* name of this device */
unsigned int caps; /* some flags about this device (SND_MIXER_CINFO_XXXX) */
int min; /* min. value when exact mode (or always 0) */
int max; /* max. value when exact mode (or always 100) */
int min_dB; /* minimum decibel value (*100) */
int max_dB; /* maximum decibel value (*100) */
int step_dB; /* step decibel value (*100) */
unsigned char reserved[16];
};
</PRE>
<HR>
</P>
<H3>int snd_mixer_channel_read( void *handle, int channel, snd_mixer_channel_t *data ) </H3>
<P>Fills the *data structure. The argument <I>channel</I> specifies
the channel (0 to N) for which is data requested. Function returns
zero if successful, otherwise it returns an error code.
<HR>
<PRE>
#define SND_MIXER_FLG_RECORD 0x00000001 /* channel record source flag */
#define SND_MIXER_FLG_MUTE_LEFT 0x00010000
#define SND_MIXER_FLG_MUTE_RIGHT 0x00020000
#define SND_MIXER_FLG_MUTE 0x00030000
#define SND_MIXER_FLG_DECIBEL 0x40000000
#define SND_MIXER_FLG_FORCE 0x80000000
struct snd_mixer_channel {
unsigned int channel; /* channel # (filled by application) */
unsigned int flags; /* some flags to read/write (SND_MIXER_FLG_XXXX) */
int left; /* min - max when exact mode (or 0 - 100) */
int right; /* min - max when exact mode (or 0 - 100) */
int left_dB; /* dB * 100 */
int right_dB; /* dB * 100 */
unsigned char reserved[16];
};
</PRE>
<HR>
</P>
<P>
<DL>
<DT><B>SND_MIXER_FLG_RECORD</B><DD><P>Record source flag.</P>
<DT><B>SND_MIXER_FLG_DECIBEL</B><DD><P>If this bit is set, driver set volume from dB variables <I>left_dB</I>
and <I>right_dB</I>.</P>
<DT><B>SND_MIXER_FLG_FORCE</B><DD><P>Force set - this bit shouldn't be used from user space. Reserved for
kernel.</P>
</DL>
</P>
<H3>int snd_mixer_channel_write( void *handle, int channel, snd_mixer_channel_t *data ) </H3>
<P>Writes the *data structure to kernel. The <I>channel</I> argument
specifies the channel (0 to N) for which is data is to be applied.
Function returns zero if successful, otherwise it returns an error code.
This functions is the opposite of <I>snd_mixer_channel_read</I>.</P>
<H3>int snd_mixer_special_read( void *handle, snd_mixer_special_t *special ) </H3>
<P>Not documented...</P>
<H3>int snd_mixer_special_write( void *handle, snd_mixer_special_t *special ) </H3>
<P>Not documented...</P>
<H3>int snd_mixer_read( void *handle, snd_mixer_callbacks_t *callbacks ) </H3>
<P>This function reads and parses data from driver. Parsed actions are returned
back to the application using the <I>callbacks</I> structure. Applications
should not parse data from the driver in standard cases. This function
returns immediately after all data is read from driver. Does not
block process.
<HR>
<PRE>
typedef struct snd_mixer_callbacks {
void *private_data; /* should be used by application */
void (*channel_was_changed)( void *private_data, int channel );
void *reserved[15]; /* reserved for future use - must be NULL!!! */
} snd_mixer_callbacks_t;
</PRE>
<HR>
</P>
<H2><A NAME="ss4.2">4.2 Examples</A></H2>
<P>The following example shows installed mixer channels for soundcard #0 and
mixer device #0 in the system, and also sets the master volume (if present)
to 50.</P>
<P>
<BLOCKQUOTE><CODE>
<HR>
<PRE>
int card = 0, device = 0, err;
void *handle;
snd_mixer_info_t info;
snd_mixer_channel_t channel;
if ( (err = snd_mixer_open( &amp;handle, card, device )) &lt; 0 ) {
fprintf( stderr, &quot;open failed: %s\n&quot;, snd_strerror( err ) );
return;
}
if ( (err = snd_mixer_info( handle, &amp;info )) &lt; 0 ) {
fprintf( stderr, &quot;info failed: %s\n&quot;, snd_strerror( err ) );
snd_mixer_close( handle );
return;
}
printf( &quot;Installed MIXER channels for card #i and device %i: %i\n&quot;,
card + 1, device, info.channels );
master = snd_mixer_channel( handle, SND_MIXER_ID_MASTER );
if ( master &gt;= 0 ) {
if ( (err = snd_mixer_read( handle, master, &amp;channel )) &lt; 0 ) {
fprintf( stderr, &quot;master read failed: %s\n&quot;, snd_strerror( err ) );
snd_mixer_close( handle );
return;
}
channel -&gt; left = channel -&gt; right = 50;
if ( (err = snd_mixer_write( handle, master, &amp;channel )) &lt; 0 ) {
fprintf( stderr, &quot;master write failed: %s\n&quot;, snd_strerror( err ) );
snd_mixer_close( handle );
return;
}
}
snd_mixer_close( handle );
</PRE>
<HR>
</CODE></BLOCKQUOTE>
</P>
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<HR>
<H2><A NAME="s5">5. Digital Audio (PCM) Interface</A></H2>
<P>Digital audio is the most commonly used method of representing sound inside
a computer. In this method sound is stored as a sequence of samples taken
from the audio signal using constant time intervals. A sample represents
volume of the signal at the moment when it was measured. In uncompressed
digital audio each sample require one or more bytes of storage. The number of
bytes required depends on number of channels (mono, stereo) and sample
format (8 or 16 bits, mu-Law, etc.). The length of this interval determines
the sampling rate. Commonly used sampling rates are between 8 kHz (telephone
quality) and 48 kHz (DAT tapes).</P>
<P>The physical devices used in digital audio are called the ADC (Analog to
Digital Converter) and DAC (Digital to Analog Converter). A device containing
both ADC and DAC is commonly known as a codec. The codec device used in
a Sound Blaster cards is called a DSP which is somewhat misleading since DSP
also stands for Digital Signal Processor (the SB DSP chip is very limited
when compared to "true" DSP chips). </P>
<P>Sampling parameters affect the quality of sound which can be reproduced from
the recorded signal. The most fundamental parameter is sampling rate which
limits the highest frequency than can be stored. It is well known (Nyquist's
Sampling Theorem) that the highest frequency that can be stored in a sampled
signal is at most 1/2 of the sampling frequency. For example, a 8 kHz sampling
rate permits the recording of a signal in which the highest frequency is less
than 4 kHz. Higher frequency signals must be filtered out before feeding them
to DAC. </P>
<P>Sample encoding limits the dynamic range of recorded signal (difference between
the faintest and the loudest signal that can be recorded). In theory the
maximum dynamic range of signal is number_of_bits * 6 dB . This means that
8 bits sampling resolution gives dynamic range of 48 dB and 16 bit resolution
gives 96 dB. </P>
<P>Quality has price. The number of bytes required to store an audio sequence
depends on sampling rate, number of channels and sampling resolution. For
example just 8000 bytes of memory is required to store one second of sound
using 8 kHz/8 bits/mono but 48 kHz/16bit/stereo takes 192 kilobytes. A 64 kbps
ISDN channel is required to transfer a 8kHz/8bit/mono audio stream in real
time, and about 1.5 Mbps is required for DAT quality (48kHz/16bit/stereo).
On the other hand it is possible to store just 5.46 seconds of sound in
a megabyte of memory when using 48kHz/16bit/stereo sampling. With
8kHz/8bits/mono it is possible to store 131 seconds of sound using the same
amount of memory. It is possible to reduce memory and communication costs by
compressing the recorded signal but this is out of the scope of this document. </P>
<H2><A NAME="ss5.1">5.1 Low-Level Layer</A></H2>
<P>Audio devices are opened exclusively for a selected direction. This doesn't
allow open from more than one processes for the same audio device in the
same direction, but does allow one open call to each playback direction and
second open call to record direction independently. Audio devices return
EBUSY error to applications when other applications have already opened the
requested direction.</P>
<P>Low-Level layer supports these formats:
<BLOCKQUOTE><CODE>
<HR>
<PRE>
#define SND_PCM_SFMT_MU_LAW 0
#define SND_PCM_SFMT_A_LAW 1
#define SND_PCM_SFMT_IMA_ADPCM 2
#define SND_PCM_SFMT_U8 3
#define SND_PCM_SFMT_S16_LE 4
#define SND_PCM_SFMT_S16_BE 5
#define SND_PCM_SFMT_S8 6
#define SND_PCM_SFMT_U16_LE 7
#define SND_PCM_SFMT_U16_BE 8
#define SND_PCM_SFMT_MPEG 9
#define SND_PCM_SFMT_GSM 10
#define SND_PCM_FMT_MU_LAW (1 &lt;&lt; SND_PCM_SFMT_MU_LAW)
#define SND_PCM_FMT_A_LAW (1 &lt;&lt; SND_PCM_SFMT_A_LAW)
#define SND_PCM_FMT_IMA_ADPCM (1 &lt;&lt; SND_PCM_SFMT_IMA_ADPCM)
#define SND_PCM_FMT_U8 (1 &lt;&lt; SND_PCM_SFMT_U8)
#define SND_PCM_FMT_S16_LE (1 &lt;&lt; SND_PCM_SFMT_S16_LE)
#define SND_PCM_FMT_S16_BE (1 &lt;&lt; SND_PCM_SFMT_S16_BE)
#define SND_PCM_FMT_S8 (1 &lt;&lt; SND_PCM_SFMT_S8)
#define SND_PCM_FMT_U16_LE (1 &lt;&lt; SND_PCM_SFMT_U16_LE)
#define SND_PCM_FMT_U16_BE (1 &lt;&lt; SND_PCM_SFMT_U16_BE)
#define SND_PCM_FMT_MPEG (1 &lt;&lt; SND_PCM_SFMT_MPEG)
#define SND_PCM_FMT_GSM (1 &lt;&lt; SND_PCM_SFMT_GSM)
</PRE>
<HR>
</CODE></BLOCKQUOTE>
Constants with prefix <B>SND_PCM_FMT_</B> are used in info structures
and constants with prefix <B>SND_PCM_SFMT_</B> are used in format structures.</P>
<H3>int snd_pcm_open( void **handle, int card, int device, int mode ) </H3>
<P>Creates a new handle and opens a connection to kernel sound
audio interface for soundcard number <I>card</I> (0-N) and audio
device number <I>device</I>. Function also checks if protocol is
compatible to prevent use of old programs with a new kernel API. Function
returns zero if successful,ful otherwise it returns an error code.
Error code -EBUSY is returned when some process ownes the selected direction.</P>
<P>Default format after opening is mono <I>mu-Law</I> at 8000Hz. This device
can be used directly for playback of standard .au (Sparc) files.</P>
<P>The following modes should be used for the <I>mode</I> argument:
<HR>
<PRE>
#define SND_PCM_OPEN_PLAYBACK (O_WRONLY)
#define SND_PCM_OPEN_RECORD (O_RDONLY)
#define SND_PCM_OPEN_DUPLEX (O_RDWR)
</PRE>
<HR>
</P>
<H3>int snd_pcm_close( void *handle ) </H3>
<P>Frees all resources allocated with audio handle and
closes the connection to the kernel sound audio interface. Function
returns zero if successful, otherwise it returns an error code.</P>
<H3>int snd_pcm_file_descriptor( void *handle ) </H3>
<P>Returns the file descriptor of the connection to the kernel sound
audio interface. Function returns an error code if an
error was encountered. </P>
<P>The file descriptor should be used for the <I>select</I> synchronous
multiplexer function for setting the read direction. Application should
call <I>snd_pcm_read</I> or <I>snd_pcm_write</I> functions if some
data is waiting for reading or a write can be performed. Calling this
function is highly recomended, as it leaves a place for the API to things
like data conversions, if needed.</P>
<H3>int snd_pcm_block_mode( void *handle, int enable ) </H3>
<P>Sets up block (default) or nonblock mode for a handle. Block mode suspends
execution of a program when <I>snd_pcm_read</I> or <I>snd_pcm_write</I>
is called for the time which is needed for the actual playback or record
over of the entire buffer. In nonblock mode, programs aren't suspended and
the above functions returns immediately with the count of bytes which were
read or written by the driver. When used in this way, don't try to use the
entire buffer after the call, but instead process the number of bytes
returned, and call the function again.</P>
<H3>int snd_pcm_info( void *handle, snd_pcm_info_t *info ) </H3>
<P>Fills the *info structure with data about the PCM device selected by
*handle. Function returns zero if successful, otherwise it returns
an error code.
<HR>
<PRE>
#define SND_PCM_INFO_CODEC 0x00000001
#define SND_PCM_INFO_DSP SND_PCM_INFO_CODEC
#define SND_PCM_INFO_MMAP 0x00000002 /* reserved */
#define SND_PCM_INFO_PLAYBACK 0x00000100
#define SND_PCM_INFO_RECORD 0x00000200
#define SND_PCM_INFO_DUPLEX 0x00000400
#define SND_PCM_INFO_DUPLEX_LIMIT 0x00000800 /* rate for playback & record are same */
struct snd_pcm_info {
unsigned int type; /* soundcard type */
unsigned int flags; /* see SND_PCM_INFO_XXXX */
unsigned char id[32]; /* ID of this PCM device */
unsigned char name[80]; /* name of this device */
unsigned char reserved[64]; /* reserved for future use */
};
</PRE>
<HR>
<DL>
<DT><B>SND_PCM_INFO_MMAP</B><DD><P>This flag is reserved and should be never used. It remains for
compatibility with Open Sound System driver.</P>
<DT><B>SND_PCM_INFO_DUPLEX_LIMIT</B><DD><P>If this bit is set, rate must be same for playback and record direction.</P>
</DL>
</P>
<H3>int snd_pcm_playback_info( void *handle, snd_pcm_playback_info_t *info ) </H3>
<P>Fills the *info structure with data about PCM playback. Function returns
zero if successful, otherwise it returns an error code.
<HR>
<PRE>
#define SND_PCM_PINFO_BATCH 0x00000001
#define SND_PCM_PINFO_8BITONLY 0x00000002
#define SND_PCM_PINFO_16BITONLY 0x00000004
struct snd_pcm_playback_info {
unsigned int flags; /* see SND_PCM_PINFO_XXXX */
unsigned int formats; /* supported formats */
unsigned int min_rate; /* min rate (in Hz) */
unsigned int max_rate; /* max rate (in Hz) */
unsigned int min_channels; /* min channels (probably always 1) */
unsigned int max_channels; /* max channels */
unsigned int buffer_size; /* playback buffer size */
unsigned int min_fragment_size; /* min fragment size in bytes */
unsigned int max_fragment_size; /* max fragment size in bytes */
unsigned int fragment_align; /* align fragment value */
unsigned char reserved[64]; /* reserved for future use */
};
</PRE>
<HR>
<DL>
<DT><B>SND_PCM_PINFO_BATCH</B><DD><P>Driver implements double buffering with this device. This means that
the chip used for data processing has its own memory, and output should be
more delayed than if a traditional codec chip is used.</P>
<DT><B>SND_PCM_PINFO_8BITONLY</B><DD><P>If this bit is set, the driver uses 8-bit format for 16-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 16-bit duplex. If this bit is set
application or highter digital audio layer should do the conversion from
16-bit samples to 8-bit samples rather than making the driver to do it in
the kernel.</P>
<DT><B>SND_PCM_PINFO_16BITONLY</B><DD><P>If this bit is set, driver uses 16-bit format for 8-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 8-bit duplex. If this bit is set the
application or highter digital audio layer should do conversion from
8-bit samples to 16-bit samples rather than making the driver to do it in
the kernel.</P>
</DL>
</P>
<H3>int snd_pcm_record_info( void *handle, snd_pcm_record_info_t *info ) </H3>
<P>Fills the *info structure. Returns zero if successful, otherwise it returns
an error code.
<HR>
<PRE>
#define SND_PCM_RINFO_BATCH 0x00000001
#define SND_PCM_RINFO_8BITONLY 0x00000002
#define SND_PCM_RINFO_16BITONLY 0x00000004
struct snd_pcm_record_info {
unsigned int flags; /* see to SND_PCM_RINFO_XXXX */
unsigned int formats; /* supported formats */
unsigned int min_rate; /* min rate (in Hz) */
unsigned int max_rate; /* max rate (in Hz) */
unsigned int min_channels; /* min channels (probably always 1) */
unsigned int max_channels; /* max channels */
unsigned int buffer_size; /* record buffer size */
unsigned int min_fragment_size; /* min fragment size in bytes */
unsigned int max_fragment_size; /* max fragment size in bytes */
unsigned int fragment_align; /* align fragment value */
unsigned char reserved[64]; /* reserved for future... */
};
</PRE>
<HR>
<DL>
<DT><B>SND_PCM_PINFO_BATCH</B><DD><P>Driver implements buffering for this device. This means that
the chip used for data processing has its own memory and output should be
more delayed than if a traditional codec chip is used.</P>
<DT><B>SND_PCM_PINFO_8BITONLY</B><DD><P>If this bit is set, the device uses 8-bit format for 16-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 16-bit duplex. If this bit is set the
application or highter digital audio layer should do conversion from
16-bit samples to 8-bit samples rather than making the driver to do it in
the kernel.</P>
<DT><B>SND_PCM_PINFO_16BITONLY</B><DD><P>If this bit is set, the device uses a 16-bit format for 8-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 8-bit duplex. If this bit is set the
application or highter digital audio layer should do the conversion from
8-bit samples to 16-bit samples rather than making the driver to do it in
the kernel.</P>
</DL>
</P>
<H3>int snd_pcm_playback_format( void *handle, snd_pcm_format_t *format ) </H3>
<P>Sets up format, rate (in Hz) and number of channels for playback, in the
desired direction. Function returns zero if successful, otherwise it
returns an error code.
<HR>
<PRE>
struct snd_pcm_format {
unsigned int format; /* SND_PCM_SFMT_XXXX */
unsigned int rate; /* rate in Hz */
unsigned int channels; /* channels (voices) */
unsigned char reserved[16];
};
</PRE>
<HR>
</P>
<H3>int snd_pcm_record_format( void *handle, snd_pcm_format_t *format ) </H3>
<P>Sets up format, rate (in Hz) and number of channels for used for recording
in the specified direction. Function returns zero if successful, otherwise
it returns an error code.
<HR>
<PRE>
struct snd_pcm_format {
unsigned int format; /* SND_PCM_SFMT_XXXX */
unsigned int rate; /* rate in Hz */
unsigned int channels; /* channels (voices) */
unsigned char reserved[16];
};
</PRE>
<HR>
</P>
<H3>int snd_pcm_playback_params( void *handle, snd_pcm_playback_params_t *params ) </H3>
<P>Sets various parameters for playback direction. Function returns zero if
successful, otherwise it returns an error code.
<HR>
<PRE>
struct snd_pcm_playback_params {
int fragment_size;
int fragments_max;
int fragments_room;
unsigned char reserved[16]; /* must be filled with zero */
};
</PRE>
<HR>
<DL>
<DT><B>fragment_size</B><DD><P>Requested size of fragment. This value should be aligned for current
format (for example to 4 if stereo 16-bit samples are used) or with the
<I>fragment_align</I> variable from <I>snd_pcm_playback_info_t</I>
structure. Its range can be from <I>min_fragment_size</I> to
<I>max_fragment_size</I>.</P>
<DT><B>fragments_max</B><DD><P>Maximum number of fragments in queue for wakeup. This number doesn't
counts partly used fragment. If current count of filled playback fragments
is greater than this value driver block application or return immediately
back if nonblock mode is active.</P>
<DT><B>fragments_room</B><DD><P>Minumum number of fragments writeable for wakeup. This value should be
in most cases 1 which means return back to application if at least
one fragment is free for playback. This value includes partly used fragments,
too.</P>
</DL>
</P>
<H3>int snd_pcm_record_params( void *handle, snd_pcm_record_params_t *params ) </H3>
<P>Function sets various parameters for the recording direction. Function returns
zero if successful, otherwise it returns an error code.
<HR>
<PRE>
struct snd_pcm_record_params {
int fragment_size;
int fragments_min;
unsigned char reserved[16];
};
</PRE>
<HR>
<DL>
<DT><B>fragment_size</B><DD><P>Requested size of fragment. This value should be aligned for current
format (for example to 4 if stereo 16-bit samples are used) or set to the
<I>fragment_align</I> variable from <I>snd_pcm_playback_info_t</I>
structure. Its range can be from <I>min_fragment_size</I> to
<I>max_fragment_size</I>.</P>
<DT><B>fragments_min</B><DD><P>Minimum filled fragments for wakeup. Driver blocks the application (if
block mode is selected) until it isn't filled with number of fragments
specified with this value.</P>
</DL>
</P>
<H3>int snd_pcm_playback_status( void *handle, snd_pcm_playback_status_t *status ) </H3>
<P>Fills the *status structure. Function returns zero if successful, otherwise
it returns an error code.
<HR>
<PRE>
struct snd_pcm_playback_status {
unsigned int rate;
int fragments;
int fragment_size;
int count;
int queue;
int underrun;
struct timeval time;
struct timeval stime;
unsigned char reserved[16];
};
</PRE>
<HR>
<DL>
<DT><B>rate</B><DD><P>Real playback rate. This value reflects hardware limitations.</P>
<DT><B>fragments</B><DD><P>Currently allocated fragments by the driver for playback direction.</P>
<DT><B>fragment_size</B><DD><P>Current fragment size used by driver for the playback direction.</P>
<DT><B>count</B><DD><P>Count of bytes writeable without blocking.</P>
<DT><B>queue</B><DD><P>Count of bytes in queue. Note: <I>(fragments * fragment_size) - queue</I>
should not be equal to <I>count</I>.</P>
<DT><B>underrun</B><DD><P>This value tells the application the number of underruns since the ast call
of <I>snd_pcm_playback_status</I>.</P>
<DT><B>time</B><DD><P>Delay till played of the first sample from next write. This value should
be used for time synchronization. Returned value is in the same format as
returned from the standard C function <I>gettimeofday( &amp;time, NULL )</I>.
This variable contains right value only if playback time mode is enabled
(look to <I>snd_pcm_playback_time</I> function).</P>
<DT><B>stime</B><DD><P>Time when playback was started.
This variable contains right value only if playback time mode is enabled
(look to <I>snd_pcm_playback_time</I> function).</P>
</DL>
</P>
<H3>int snd_pcm_record_status( void *handle, snd_pcm_record_status_t *status ) </H3>
<P>Fills the *status structure. Function returns zero if successful, otherwise
it returns an error code.
<HR>
<PRE>
struct snd_pcm_record_status {
unsigned int rate;
int fragments;
int fragment_size;
int count;
int free;
int overrun;
struct timeval time;
unsigned char reserved[16];
};
</PRE>
<HR>
<DL>
<DT><B>rate</B><DD><P>Real record rate. This value reflects hardware limitations.</P>
<DT><B>fragments</B><DD><P>Currently allocated fragments by driver for the record direction.</P>
<DT><B>fragment_size</B><DD><P>Current fragment size used by driver for the record direction.</P>
<DT><B>count</B><DD><P>Count of bytes readable without blocking.</P>
<DT><B>free</B><DD><P>Count of bytes in buffer still free. Note: <I>(fragments * fragment_size) - free</I>
should not be equal to <I>count</I>.</P>
<DT><B>overrun</B><DD><P>This value tells application the count of overruns since the last call
to <I>snd_pcm_record_status</I>.</P>
<DT><B>time</B><DD><P>Lag since the next sample read was recorded. This value should be used for time
synchronization. Returned value is in the same format as returned by the
from standard C function <I>gettimeofday( &amp;time, NULL )</I>. This
variable contains right value only if record time mode is enabled (look to
<I>snd_pcm_record_time</I> function).</P>
<DT><B>stime</B><DD><P>Time when record was started. This variable contains right value only if
record time mode is enabled (look to <I>snd_pcm_record_time</I> function).</P>
</DL>
</P>
<H3>int snd_pcm_drain_playback( void *handle ) </H3>
<P>This function drain playback buffers immediately. Function returns zero
if successful, otherwise it returns an error code. </P>
<H3>int snd_pcm_flush_playback( void *handle ) </H3>
<P>This function flushes the playback buffers. It blocks the program while the
all the waiting samples in kernel playback buffers are processed. Function
returns zero if successful, otherwise it returns an error code.</P>
<H3>int snd_pcm_flush_record( void *handle ) </H3>
<P>This function flushes (destroyes) record buffers. Function returns zero
if successful, otherwise it returns an error code. </P>
<H3>int snd_pcm_playback_time( void *handle, int enable ) </H3>
<P>This function enables or disables time mode for playback direction. Time mode
allows to application better time synchronization. Function returns zero
if successful, otherwise it returns an error code.</P>
<H3>int snd_pcm_record_time( void *handle, int enable ) </H3>
<P>This function enables or disables time mode for record direction. Time mode
allows to application better time synchronization. Function returns zero
if successful, otherwise it returns an error code.</P>
<H3>ssize_t snd_pcm_write( void *handle, const void *buffer, size_t size ) </H3>
<P>Writes samples to the device which must be in the proper format
specified by the <I>snd_pcm_playback_format</I> function. Function
returns zero or positive value if playback was successful (value represents
count of bytes which was successfuly written to device) or an
error value if error occured. Function should suspend process if
block mode is active.</P>
<H3>ssize_t snd_pcm_read( void *handle, void *buffer, size_t size ) </H3>
<P>Function reads samples from driver. Samples are in format specified
by <I>snd_pcm_record_format</I> function. Function returns zero
or positive value if record was success (value represents count of bytes
which was successfuly read from device) or negative error value if
error occured. Function should suspend process if block mode is active.</P>
<H2><A NAME="ss5.2">5.2 Examples</A></H2>
<P>The following example shows how to play the first 512kB from the
/tmp/test.au file with soundcard #0 and PCM device #0:</P>
<P>
<BLOCKQUOTE><CODE>
<HR>
<PRE>
int card = 0, device = 0, err, fd, count, size, idx;
void *handle;
snd_pcm_format_t format;
char *buffer;
buffer = (char *)malloc( 512 * 1024 );
if ( !buffer ) return;
if ( (err = snd_pcm_open( &amp;handle, card, device, SND_PCM_OPEN_PLAYBACK )) &lt; 0 ) {
fprintf( stderr, &quot;open failed: %s\n&quot;, snd_strerror( err ) );
return;
}
format.format = SND_PCM_SFMT_MU_LAW;
format.rate = 8000;
format.voices = 1;
if ( (err = snd_pcm_playback_format( handle, &amp;format )) &lt; 0 ) {
fprintf( stderr, &quot;format setup failed: %s\n&quot;, snd_strerror( err ) );
snd_pcm_close( handle );
return;
}
fd = open( &quot;/tmp/test.au&quot; );
if ( fd &lt; 0 ) {
perror( &quot;open file&quot; );
snd_pcm_close( handle );
return;
}
idx = 0;
count = read( fd, buffer, 512 * 1024 );
if ( count &lt;= 0 ) {
perror( &quot;read from file&quot; );
snd_pcm_close( handle );
return;
}
close( fd );
if ( !memcmp( buffer, &quot;.snd&quot;, 4 ) ) {
idx = (buffer[4]&lt;&lt;24)|(buffer[5]&lt;&lt;16)|(buffer[6]&lt;&lt;8)|(buffer[7]);
if ( idx &gt; 128 ) idx = 128;
if ( idx &gt; count ) idx = count;
}
size = snd_pcm_write( handle, &amp;buffer[ idx ], count - idx );
printf( &quot;Bytes written %i from %i...\n&quot;, size, count - idx );
snd_pcm_close( handle );
free( buffer );
</PRE>
<HR>
</CODE></BLOCKQUOTE>
</P>
<HR>
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<HR>
<H1>Advanced Linux Sound Architecture - Library API</H1>
<H2>Jaroslav Kysela <CODE>&lt;perex@jcu.cz&gt;</CODE> with assistance from Alan Robinson</H2>v0.0.3, 25 March 1998
<P><HR><EM>This document describes, in full detail, the Advanced Linux Sound Architecture library API.</EM><HR></P>
<P>
<H2><A NAME="toc1">1.</A> <A HREF="soundapi-1.html">Introduction</A></H2>
<P>
<H2><A NAME="toc2">2.</A> <A HREF="soundapi-2.html">Error Codes</A></H2>
<UL>
<LI><A HREF="soundapi-2.html#ss2.1">2.1 Error Codes in Detail</A>
<LI><A HREF="soundapi-2.html#ss2.2">2.2 Functions</A>
</UL>
<P>
<H2><A NAME="toc3">3.</A> <A HREF="soundapi-3.html">Control Interface</A></H2>
<UL>
<LI><A HREF="soundapi-3.html#ss3.1">3.1 Low-Level Layer</A>
<LI><A HREF="soundapi-3.html#ss3.2">3.2 Examples</A>
</UL>
<P>
<H2><A NAME="toc4">4.</A> <A HREF="soundapi-4.html">Mixer Interface</A></H2>
<UL>
<LI><A HREF="soundapi-4.html#ss4.1">4.1 Low-Level Layer</A>
<LI><A HREF="soundapi-4.html#ss4.2">4.2 Examples</A>
</UL>
<P>
<H2><A NAME="toc5">5.</A> <A HREF="soundapi-5.html">Digital Audio (PCM) Interface</A></H2>
<UL>
<LI><A HREF="soundapi-5.html#ss5.1">5.1 Low-Level Layer</A>
<LI><A HREF="soundapi-5.html#ss5.2">5.2 Examples</A>
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<!doctype linuxdoc system>
<!-- Advanced Linux Sound Architecture - Library API -->
<article>
<!-- Title information -->
<title>Advanced Linux Sound Architecture - Library API
<author>Jaroslav Kysela <tt>&lt;perex@jcu.cz&gt;</tt> with assistance from Alan Robinson
<date>v0.0.3, 25 March 1998
<abstract>
This document describes, in full detail, the Advanced Linux Sound
Architecture library API.
</abstract>
<!-- Table of contents -->
<toc>
<!-- Begin the document -->
<sect>Introduction
<p>
The Advanced Linux Sound Architecture comes with a kernel API & library API.
This document describes the library API and how it interfaces with the kernel
API. The kernal API will probably never be documented in standalone form.
<P>
Application programmers should use the library API rather than kernel API.
The Library offers 100% of the functionally of the kernel API, but add next
major improvements in usability, making the application code simpler and
better looking. In addition, some of the some fixes/compatibility code in,
may be placed in the library code instead of the kernel driver.
<p>
For a complete list of all variables and functions in the API you should look
at the following header files:
<enum>
<item>/usr/include/sys/soundlib.h
<item>/usr/include/linux/sound.h
<item>/usr/include/linux/sounddetect.h
</enum>
<sect>Error Codes
<p>
All functions return int (or some sort of signed value). If this value
is negative it represents an error code. Codes up to SND_ERROR_BEGIN (500000)
represents standard system errors. Codes equal or greather than this value
represents sound library API errors. All error codes begin with the prefix
<it>SND_ERROR_</it>.
<sect1>Error Codes in Detail
<p>
<descrip>
<tag>SND_ERROR_UNCOMPATIBLE_VERSION (500000)</tag>
This error is caused if the driver uses an incompatible kernel API for this
interface and hence the library doesn't know how this API can be used.
</descrip>
<sect1>Functions
<p>
<sect2>const char *snd_strerror( int errnum )
<p>
This functions converts error code to a string. Its functionality is the same
as the <it>strerror</it> function from the standard C library, but this
function returns correct strings for sound error codes, too.
<sect>Control Interface
<p>
The control interfaces gives application various information about the
currently installed sound driver in the system. The interface should be used
to detect if another sound interface is present for selected soundcard or,
for example, to create a list of devices (MIXER, PCM etc) from which
the user can select.
<sect1>Low-Level Layer
<sect2>int snd_cards( void )
<p>
Returns the number of soundcards present in the system, if any. Otherwise
it returns a negative value, which maps to an error code. This function
will return 0 if no soundcards are detected.
<sect2>unsigned int snd_cards_mask( void )
<p>
Returns the bitmap of soundcards present in the system, if any. Otherwise
it returns a negative value, which maps to an error code. This function
will return 0 if no soundcards are detected. First soundcard is represented
with bit 0.
<sect2>int snd_ctl_open( void **handle, int card )
<p>
Creates a new handle and opens communication with the kernel sound
control interface for soundcard number <it>card</it> (0-N). The function
also checks if protocol is compatible, so as to prevent the use of old
programs with a new kernel API. Function returns zero if successful,
otherwise an error code is returned.
<sect2>int snd_ctl_close( void *handle )
<p>
Function frees all resources allocated with control handle and
closes the kernel sound control interface. Function returns zero if
successful, otherwise it returns an error code.
<sect2>int snd_ctl_file_descriptor( void *handle )
<p>
Function returns file descriptor for the kernel sound control interface.
This function should be used in very special cases. Function returns
a negative error code if some error was encountered.
<sect2>int snd_ctl_hw_info( void *handle, struct snd_ctl_hw_info *info )
<p>
Fills the info structure with data about the sound hardware referenced
by handle. Function returns zero if successful, otherwise it returns
an error code.
<code>
#define SND_CTL_GCAPS_MIDI 0x0000001 /* driver has MIDI interface */
#define SND_CTL_LCAPS_SYNTH 0x0000001 /* soundcard has synthesizer */
#define SND_CTL_LCAPS_RAWFM 0x0000002 /* soundcard has RAW FM/OPL3 */
struct snd_ctl_hw_info {
unsigned int type; /* type of card - see SND_CARD_TYPE_XXXX */
unsigned int gcaps; /* see SND_CTL_GCAPS_XXXX */
unsigned int lcaps; /* see SND_CTL_LCAPS_XXXX */
unsigned int pcmdevs; /* count of PCM devices (0 to N) */
unsigned int mixerdevs; /* count of MIXER devices (0 to N) */
unsigned int mididevs; /* count of raw MIDI devices (0 to N) */
char id[8]; /* ID of card (user selectable) */
char name[80]; /* name/info text about soundcard */
unsigned char reserved[128]; /* reserved for future use */
};
</code>
<sect2>int snd_ctl_pcm_info( void *handle, int dev, snd_pcm_info_t *info )
<p>
Fills the *info structure with data about the PCM device. Function returns
zero if successful, otherwise it returns an error code. Details about
the snd_pcm_info_t structure are in the <bf>Digital Audio (PCM) Interface</bf>
section. The argument <it>dev</it> selects the device number for the
soundcard referenced by *handle. Its range is 0 to N where N is
<it>struct snd_ctl_hw_info -> pcmdevs - 1</it>. This function will work if
the selected PCM device is busy, too. It should be used to collect
information about PCM devices without exclusive lock.
<sect2>int snd_ctl_pcm_playback_info( void *handle, int dev, snd_pcm_playback_info_t *info )
<p>
Fills the *info structure with data about the PCM device and playback direction.
Function returns zero if successful, otherwise it returns an error code.
Details about the snd_pcm_playback_info_t structure are in the
<bf>Digital Audio (PCM) Interface</bf> section. The argument <it>dev</it>
selects the device number for the soundcard referenced by *handle. Its
range is 0 to N where N is <it>struct snd_ctl_hw_info -> pcmdevs - 1</it>.
This function will work if the selected PCM device is busy, too. It should
be used to collect information about PCM devices without exclusive lock.
<sect2>int snd_ctl_pcm_record_info( void *handle, int dev, snd_pcm_record_info_t *info )
<p>
Fills the *info structure with data about the PCM device and record direction.
Function returns zero if successful, otherwise it returns an error code.
Details about the snd_pcm_record_info_t structure are in the
<bf>Digital Audio (PCM) Interface</bf> section. The argument <it>dev</it>
selects the device number for the soundcard referenced by *handle. Its
range is 0 to N where N is <it>struct snd_ctl_hw_info -> pcmdevs - 1</it>.
This function will work if the selected PCM device is busy, too. It should
be used to collect information about PCM devices without exclusive lock.
<sect2>int snd_ctl_mixer_info( void *handle, int dev, snd_mixer_info_t *info )
<p>
Fills the *info structure with data about the mixer device. Returns zero
if successful, otherwise it returns an error code. Details about the
snd_mixer_info_t structure are in the <bf>Mixer Interface</bf> section.
The argument <it>dev</it> specifies the device number for the appropriate
soundcard. Its range is 0 to N where N found from
<it>struct snd_ctl_hw_info -> mixerdevs - 1</it>.
It should be used to collect information about mixer devices.
<sect1>Examples
<p>
The following example shows how all PCM devices can be detected for the first
soundcard (#0) in the system.
<tscreen><code>
int card = 0, err;
void *handle;
stuct snd_ctl_hw_info info;
if ( (err = snd_ctl_open( &ero;handle, card )) < 0 ) {
fprintf( stderr, "open failed: %s\n", snd_strerror( err ) );
return;
}
if ( (err = snd_ctl_hw_info( handle, &ero;info )) < 0 ) {
fprintf( stderr, "hw info failed: %s\n", snd_strerror( err ) );
snd_ctl_close( handle );
return;
}
printf( "Installed PCM devices for card #i: %i\n", card + 1, info.pcmdevs );
snd_ctl_close( handle );
</code></tscreen>
<sect>Mixer Interface
<p>
The Mixer Interface allows applications to change the volume level of
a soundcard's input/output channels in both the linear range (0-100)
and in decibels. It also supports features like hardware mute, input
sound source, etc.
<sect1>Low-Level Layer
<p>
Mixer devices aren't opened exclusively. This allows applications to
open a device multiple times with one or more processes.
<sect2>int snd_mixer_open( void **handle, int card, int device )
<p>
Creates new handle and opens a connection to the kernel sound
mixer interface for soundcard number <it>card</it> (0-N) and mixer
device number <it>device</it>. Also checks if protocol is
compatible to prevent use of old programs with new kernel API. Function
returns zero if successful, otherwise it returns an error code.
<sect2>int snd_mixer_close( void *handle )
<p>
Frees all resources allocated to the mixer handle and
closes its connection to the kernel sound mixer interface. Function
returns zero if successful, otherwise it returns an error code.
<sect2>int snd_mixer_file_descriptor( void *handle )
<p>
Returns the file descriptor for the connection to the kernel sound
mixer interface. This function should be used only in very
special cases. Function returns a negative error code if an
error was encountered.
<p>
The file descriptor should be used for the <it>select</it> synchronous
multiplexer function for deterimeing read direction. Applications should
call <it>snd_mixer_read</it> function if some data is waiting to be read.
It is recomended that you do this, since it leaves place for this function
to handle some new kernel API specifications.
<sect2>int snd_mixer_channels( void *handle )
<p>
Returns the count of mixer channels for appropriate mixer device, otherwise
the return value is negative, and signifies an error code. Never returns
zero.
<sect2>int snd_mixer_info( void *handle, snd_mixer_info_t *info )
<p>
Fills the *info structure with information about the mixer associated with
*handle. Returns zero if successful, otherwise it returns an error code.
<code>
#define SND_MIXER_INFO_CAP_EXCL_RECORD 0x00000001
struct snd_mixer_info {
unsigned int type; /* type of soundcard - SND_CARD_TYPE_XXXX */
unsigned int channels; /* count of mixer devices */
unsigned int caps; /* some flags about this device (SND_MIXER_INFO_CAP_XXXX) */
unsigned char id[32]; /* ID of this mixer */
unsigned char name[80]; /* name of this device */
char reserved[ 32 ]; /* reserved for future use */
};
</code>
<sect2>int snd_mixer_channel( void *handle, const char *channel_id )
<p>
Returns the channel number (index) associated with channel_id (channel name),
or returns an error code.
<code>
#define SND_MIXER_ID_MASTER "Master"
#define SND_MIXER_ID_BASS "Bass"
#define SND_MIXER_ID_TREBLE "Treble"
#define SND_MIXER_ID_SYNTHESIZER "Synth"
#define SND_MIXER_ID_SYNTHESIZER1 "Synth 1"
#define SND_MIXER_ID_FM "FM"
#define SND_MIXER_ID_EFFECT "Effect"
#define SND_MIXER_ID_PCM "PCM"
#define SND_MIXER_ID_PCM1 "PCM 1"
#define SND_MIXER_ID_LINE "Line-In"
#define SND_MIXER_ID_MIC "MIC"
#define SND_MIXER_ID_CD "CD"
#define SND_MIXER_ID_GAIN "Record-Gain"
#define SND_MIXER_ID_IGAIN "In-Gain"
#define SND_MIXER_ID_OGAIN "Out-Gain"
#define SND_MIXER_ID_LOOPBACK "Loopback"
#define SND_MIXER_ID_SPEAKER "PC Speaker"
#define SND_MIXER_ID_AUXA "Aux A"
#define SND_MIXER_ID_AUXB "Aux B"
#define SND_MIXER_ID_AUXC "Aux C"
</code>
<sect2>int snd_mixer_exact_mode( void *handle, int enable )
<p>
Turns on or off (by default) exact mode. This mode allows to application
set/get volume values in exact range which uses hardware. In non-exact
mode is range always from 0 to 100 and conversion to hardware range does
driver. Function returns zero if successful, otherwise it returns an error
code.
<sect2>int snd_mixer_channel_info( void *handle, int channel, snd_mixer_channel_info_t *info )
<p>
Fills the *info structure. The argument <it>channel</it> specifies channel
(0 to N) for which is the info requested. Function returns zero if
successful, otherwise it returns an error code.
<code>
#define SND_MIXER_CINFO_CAP_RECORD 0x00000001
#define SND_MIXER_CINFO_CAP_STEREO 0x00000002
#define SND_MIXER_CINFO_CAP_MUTE 0x00000004
#define SND_MIXER_CINFO_CAP_HWMUTE 0x00000008 /* channel supports hardware mute */
#define SND_MIXER_CINFO_CAP_DIGITAL 0x00000010 /* channel does digital (not analog) mixing */
#define SND_MIXER_CINOF_CAP_INPUT 0x00000020 /* input channel */
struct snd_mixer_channel_info {
unsigned int channel; /* channel # (filled by application) */
unsigned int parent; /* parent channel # or SND_MIXER_PARENT */
unsigned char name[12]; /* name of this device */
unsigned int caps; /* some flags about this device (SND_MIXER_CINFO_XXXX) */
int min; /* min. value when exact mode (or always 0) */
int max; /* max. value when exact mode (or always 100) */
int min_dB; /* minimum decibel value (*100) */
int max_dB; /* maximum decibel value (*100) */
int step_dB; /* step decibel value (*100) */
unsigned char reserved[16];
};
</code>
<sect2>int snd_mixer_channel_read( void *handle, int channel, snd_mixer_channel_t *data )
<p>
Fills the *data structure. The argument <it>channel</it> specifies
the channel (0 to N) for which is data requested. Function returns
zero if successful, otherwise it returns an error code.
<code>
#define SND_MIXER_FLG_RECORD 0x00000001 /* channel record source flag */
#define SND_MIXER_FLG_MUTE_LEFT 0x00010000
#define SND_MIXER_FLG_MUTE_RIGHT 0x00020000
#define SND_MIXER_FLG_MUTE 0x00030000
#define SND_MIXER_FLG_DECIBEL 0x40000000
#define SND_MIXER_FLG_FORCE 0x80000000
struct snd_mixer_channel {
unsigned int channel; /* channel # (filled by application) */
unsigned int flags; /* some flags to read/write (SND_MIXER_FLG_XXXX) */
int left; /* min - max when exact mode (or 0 - 100) */
int right; /* min - max when exact mode (or 0 - 100) */
int left_dB; /* dB * 100 */
int right_dB; /* dB * 100 */
unsigned char reserved[16];
};
</code>
<p>
<descrip>
<tag>SND_MIXER_FLG_RECORD</tag>
Record source flag.
<tag>SND_MIXER_FLG_DECIBEL</tag>
If this bit is set, driver set volume from dB variables <it>left_dB</it>
and <it>right_dB</it>.
<tag>SND_MIXER_FLG_FORCE</tag>
Force set - this bit shouldn't be used from user space. Reserved for
kernel.
</descrip>
<sect2>int snd_mixer_channel_write( void *handle, int channel, snd_mixer_channel_t *data )
<p>
Writes the *data structure to kernel. The <it>channel</it> argument
specifies the channel (0 to N) for which is data is to be applied.
Function returns zero if successful, otherwise it returns an error code.
This functions is the opposite of <it>snd_mixer_channel_read</it>.
<sect2>int snd_mixer_special_read( void *handle, snd_mixer_special_t *special )
<p>
Not documented...
<sect2>int snd_mixer_special_write( void *handle, snd_mixer_special_t *special )
<p>
Not documented...
<sect2>int snd_mixer_read( void *handle, snd_mixer_callbacks_t *callbacks )
<p>
This function reads and parses data from driver. Parsed actions are returned
back to the application using the <it>callbacks</it> structure. Applications
should not parse data from the driver in standard cases. This function
returns immediately after all data is read from driver. Does not
block process.
<code>
typedef struct snd_mixer_callbacks {
void *private_data; /* should be used by application */
void (*channel_was_changed)( void *private_data, int channel );
void *reserved[15]; /* reserved for future use - must be NULL!!! */
} snd_mixer_callbacks_t;
</code>
<sect1>Examples
<p>
The following example shows installed mixer channels for soundcard #0 and
mixer device #0 in the system, and also sets the master volume (if present)
to 50.
<tscreen><code>
int card = 0, device = 0, err;
void *handle;
snd_mixer_info_t info;
snd_mixer_channel_t channel;
if ( (err = snd_mixer_open( &ero;handle, card, device )) < 0 ) {
fprintf( stderr, "open failed: %s\n", snd_strerror( err ) );
return;
}
if ( (err = snd_mixer_info( handle, &ero;info )) < 0 ) {
fprintf( stderr, "info failed: %s\n", snd_strerror( err ) );
snd_mixer_close( handle );
return;
}
printf( "Installed MIXER channels for card #i and device %i: %i\n",
card + 1, device, info.channels );
master = snd_mixer_channel( handle, SND_MIXER_ID_MASTER );
if ( master >= 0 ) {
if ( (err = snd_mixer_read( handle, master, &ero;channel )) < 0 ) {
fprintf( stderr, "master read failed: %s\n", snd_strerror( err ) );
snd_mixer_close( handle );
return;
}
channel -> left = channel -> right = 50;
if ( (err = snd_mixer_write( handle, master, &ero;channel )) < 0 ) {
fprintf( stderr, "master write failed: %s\n", snd_strerror( err ) );
snd_mixer_close( handle );
return;
}
}
snd_mixer_close( handle );
</code></tscreen>
<sect>Digital Audio (PCM) Interface
<p>
Digital audio is the most commonly used method of representing sound inside
a computer. In this method sound is stored as a sequence of samples taken
from the audio signal using constant time intervals. A sample represents
volume of the signal at the moment when it was measured. In uncompressed
digital audio each sample require one or more bytes of storage. The number of
bytes required depends on number of channels (mono, stereo) and sample
format (8 or 16 bits, mu-Law, etc.). The length of this interval determines
the sampling rate. Commonly used sampling rates are between 8 kHz (telephone
quality) and 48 kHz (DAT tapes).
<p>
The physical devices used in digital audio are called the ADC (Analog to
Digital Converter) and DAC (Digital to Analog Converter). A device containing
both ADC and DAC is commonly known as a codec. The codec device used in
a Sound Blaster cards is called a DSP which is somewhat misleading since DSP
also stands for Digital Signal Processor (the SB DSP chip is very limited
when compared to "true" DSP chips).
<p>
Sampling parameters affect the quality of sound which can be reproduced from
the recorded signal. The most fundamental parameter is sampling rate which
limits the highest frequency than can be stored. It is well known (Nyquist's
Sampling Theorem) that the highest frequency that can be stored in a sampled
signal is at most 1/2 of the sampling frequency. For example, a 8 kHz sampling
rate permits the recording of a signal in which the highest frequency is less
than 4 kHz. Higher frequency signals must be filtered out before feeding them
to DAC.
<p>
Sample encoding limits the dynamic range of recorded signal (difference between
the faintest and the loudest signal that can be recorded). In theory the
maximum dynamic range of signal is number_of_bits * 6 dB . This means that
8 bits sampling resolution gives dynamic range of 48 dB and 16 bit resolution
gives 96 dB.
<p>
Quality has price. The number of bytes required to store an audio sequence
depends on sampling rate, number of channels and sampling resolution. For
example just 8000 bytes of memory is required to store one second of sound
using 8 kHz/8 bits/mono but 48 kHz/16bit/stereo takes 192 kilobytes. A 64 kbps
ISDN channel is required to transfer a 8kHz/8bit/mono audio stream in real
time, and about 1.5 Mbps is required for DAT quality (48kHz/16bit/stereo).
On the other hand it is possible to store just 5.46 seconds of sound in
a megabyte of memory when using 48kHz/16bit/stereo sampling. With
8kHz/8bits/mono it is possible to store 131 seconds of sound using the same
amount of memory. It is possible to reduce memory and communication costs by
compressing the recorded signal but this is out of the scope of this document.
<sect1>Low-Level Layer
<p>
Audio devices are opened exclusively for a selected direction. This doesn't
allow open from more than one processes for the same audio device in the
same direction, but does allow one open call to each playback direction and
second open call to record direction independently. Audio devices return
EBUSY error to applications when other applications have already opened the
requested direction.
<p>
Low-Level layer supports these formats:
<tscreen><code>
#define SND_PCM_SFMT_MU_LAW 0
#define SND_PCM_SFMT_A_LAW 1
#define SND_PCM_SFMT_IMA_ADPCM 2
#define SND_PCM_SFMT_U8 3
#define SND_PCM_SFMT_S16_LE 4
#define SND_PCM_SFMT_S16_BE 5
#define SND_PCM_SFMT_S8 6
#define SND_PCM_SFMT_U16_LE 7
#define SND_PCM_SFMT_U16_BE 8
#define SND_PCM_SFMT_MPEG 9
#define SND_PCM_SFMT_GSM 10
#define SND_PCM_FMT_MU_LAW (1 << SND_PCM_SFMT_MU_LAW)
#define SND_PCM_FMT_A_LAW (1 << SND_PCM_SFMT_A_LAW)
#define SND_PCM_FMT_IMA_ADPCM (1 << SND_PCM_SFMT_IMA_ADPCM)
#define SND_PCM_FMT_U8 (1 << SND_PCM_SFMT_U8)
#define SND_PCM_FMT_S16_LE (1 << SND_PCM_SFMT_S16_LE)
#define SND_PCM_FMT_S16_BE (1 << SND_PCM_SFMT_S16_BE)
#define SND_PCM_FMT_S8 (1 << SND_PCM_SFMT_S8)
#define SND_PCM_FMT_U16_LE (1 << SND_PCM_SFMT_U16_LE)
#define SND_PCM_FMT_U16_BE (1 << SND_PCM_SFMT_U16_BE)
#define SND_PCM_FMT_MPEG (1 << SND_PCM_SFMT_MPEG)
#define SND_PCM_FMT_GSM (1 << SND_PCM_SFMT_GSM)
</code></tscreen>
Constants with prefix <bf>SND_PCM_FMT_</bf> are used in info structures
and constants with prefix <bf>SND_PCM_SFMT_</bf> are used in format structures.
<sect2>int snd_pcm_open( void **handle, int card, int device, int mode )
<p>
Creates a new handle and opens a connection to kernel sound
audio interface for soundcard number <it>card</it> (0-N) and audio
device number <it>device</it>. Function also checks if protocol is
compatible to prevent use of old programs with a new kernel API. Function
returns zero if successful,ful otherwise it returns an error code.
Error code -EBUSY is returned when some process ownes the selected direction.
<p>
Default format after opening is mono <it>mu-Law</it> at 8000Hz. This device
can be used directly for playback of standard .au (Sparc) files.
<p>
The following modes should be used for the <it>mode</it> argument:
<code>
#define SND_PCM_OPEN_PLAYBACK (O_WRONLY)
#define SND_PCM_OPEN_RECORD (O_RDONLY)
#define SND_PCM_OPEN_DUPLEX (O_RDWR)
</code>
<sect2>int snd_pcm_close( void *handle )
<p>
Frees all resources allocated with audio handle and
closes the connection to the kernel sound audio interface. Function
returns zero if successful, otherwise it returns an error code.
<sect2>int snd_pcm_file_descriptor( void *handle )
<p>
Returns the file descriptor of the connection to the kernel sound
audio interface. Function returns an error code if an
error was encountered.
<p>
The file descriptor should be used for the <it>select</it> synchronous
multiplexer function for setting the read direction. Application should
call <it>snd_pcm_read</it> or <it>snd_pcm_write</it> functions if some
data is waiting for reading or a write can be performed. Calling this
function is highly recomended, as it leaves a place for the API to things
like data conversions, if needed.
<sect2>int snd_pcm_block_mode( void *handle, int enable )
<p>
Sets up block (default) or nonblock mode for a handle. Block mode suspends
execution of a program when <it>snd_pcm_read</it> or <it>snd_pcm_write</it>
is called for the time which is needed for the actual playback or record
over of the entire buffer. In nonblock mode, programs aren't suspended and
the above functions returns immediately with the count of bytes which were
read or written by the driver. When used in this way, don't try to use the
entire buffer after the call, but instead process the number of bytes
returned, and call the function again.
<sect2>int snd_pcm_info( void *handle, snd_pcm_info_t *info )
<p>
Fills the *info structure with data about the PCM device selected by
*handle. Function returns zero if successful, otherwise it returns
an error code.
<code>
#define SND_PCM_INFO_CODEC 0x00000001
#define SND_PCM_INFO_DSP SND_PCM_INFO_CODEC
#define SND_PCM_INFO_MMAP 0x00000002 /* reserved */
#define SND_PCM_INFO_PLAYBACK 0x00000100
#define SND_PCM_INFO_RECORD 0x00000200
#define SND_PCM_INFO_DUPLEX 0x00000400
#define SND_PCM_INFO_DUPLEX_LIMIT 0x00000800 /* rate for playback & record are same */
struct snd_pcm_info {
unsigned int type; /* soundcard type */
unsigned int flags; /* see SND_PCM_INFO_XXXX */
unsigned char id[32]; /* ID of this PCM device */
unsigned char name[80]; /* name of this device */
unsigned char reserved[64]; /* reserved for future use */
};
</code>
<descrip>
<tag>SND_PCM_INFO_MMAP</tag>
This flag is reserved and should be never used. It remains for
compatibility with Open Sound System driver.
<tag>SND_PCM_INFO_DUPLEX_LIMIT</tag>
If this bit is set, rate must be same for playback and record direction.
</descrip>
<sect2>int snd_pcm_playback_info( void *handle, snd_pcm_playback_info_t *info )
<p>
Fills the *info structure with data about PCM playback. Function returns
zero if successful, otherwise it returns an error code.
<code>
#define SND_PCM_PINFO_BATCH 0x00000001
#define SND_PCM_PINFO_8BITONLY 0x00000002
#define SND_PCM_PINFO_16BITONLY 0x00000004
struct snd_pcm_playback_info {
unsigned int flags; /* see SND_PCM_PINFO_XXXX */
unsigned int formats; /* supported formats */
unsigned int min_rate; /* min rate (in Hz) */
unsigned int max_rate; /* max rate (in Hz) */
unsigned int min_channels; /* min channels (probably always 1) */
unsigned int max_channels; /* max channels */
unsigned int buffer_size; /* playback buffer size */
unsigned int min_fragment_size; /* min fragment size in bytes */
unsigned int max_fragment_size; /* max fragment size in bytes */
unsigned int fragment_align; /* align fragment value */
unsigned char reserved[64]; /* reserved for future use */
};
</code>
<descrip>
<tag>SND_PCM_PINFO_BATCH</tag>
Driver implements double buffering with this device. This means that
the chip used for data processing has its own memory, and output should be
more delayed than if a traditional codec chip is used.
<tag>SND_PCM_PINFO_8BITONLY</tag>
If this bit is set, the driver uses 8-bit format for 16-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 16-bit duplex. If this bit is set
application or highter digital audio layer should do the conversion from
16-bit samples to 8-bit samples rather than making the driver to do it in
the kernel.
<tag>SND_PCM_PINFO_16BITONLY</tag>
If this bit is set, driver uses 16-bit format for 8-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 8-bit duplex. If this bit is set the
application or highter digital audio layer should do conversion from
8-bit samples to 16-bit samples rather than making the driver to do it in
the kernel.
</descrip>
<sect2>int snd_pcm_record_info( void *handle, snd_pcm_record_info_t *info )
<p>
Fills the *info structure. Returns zero if successful, otherwise it returns
an error code.
<code>
#define SND_PCM_RINFO_BATCH 0x00000001
#define SND_PCM_RINFO_8BITONLY 0x00000002
#define SND_PCM_RINFO_16BITONLY 0x00000004
struct snd_pcm_record_info {
unsigned int flags; /* see to SND_PCM_RINFO_XXXX */
unsigned int formats; /* supported formats */
unsigned int min_rate; /* min rate (in Hz) */
unsigned int max_rate; /* max rate (in Hz) */
unsigned int min_channels; /* min channels (probably always 1) */
unsigned int max_channels; /* max channels */
unsigned int buffer_size; /* record buffer size */
unsigned int min_fragment_size; /* min fragment size in bytes */
unsigned int max_fragment_size; /* max fragment size in bytes */
unsigned int fragment_align; /* align fragment value */
unsigned char reserved[64]; /* reserved for future... */
};
</code>
<descrip>
<tag>SND_PCM_PINFO_BATCH</tag>
Driver implements buffering for this device. This means that
the chip used for data processing has its own memory and output should be
more delayed than if a traditional codec chip is used.
<tag>SND_PCM_PINFO_8BITONLY</tag>
If this bit is set, the device uses 8-bit format for 16-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 16-bit duplex. If this bit is set the
application or highter digital audio layer should do conversion from
16-bit samples to 8-bit samples rather than making the driver to do it in
the kernel.
<tag>SND_PCM_PINFO_16BITONLY</tag>
If this bit is set, the device uses a 16-bit format for 8-bit samples and
does software conversion. This bit is set on broken SoundBlaster 16/AWE
soundcards which can't do full 8-bit duplex. If this bit is set the
application or highter digital audio layer should do the conversion from
8-bit samples to 16-bit samples rather than making the driver to do it in
the kernel.
</descrip>
<sect2>int snd_pcm_playback_format( void *handle, snd_pcm_format_t *format )
<p>
Sets up format, rate (in Hz) and number of channels for playback, in the
desired direction. Function returns zero if successful, otherwise it
returns an error code.
<code>
struct snd_pcm_format {
unsigned int format; /* SND_PCM_SFMT_XXXX */
unsigned int rate; /* rate in Hz */
unsigned int channels; /* channels (voices) */
unsigned char reserved[16];
};
</code>
<sect2>int snd_pcm_record_format( void *handle, snd_pcm_format_t *format )
<p>
Sets up format, rate (in Hz) and number of channels for used for recording
in the specified direction. Function returns zero if successful, otherwise
it returns an error code.
<code>
struct snd_pcm_format {
unsigned int format; /* SND_PCM_SFMT_XXXX */
unsigned int rate; /* rate in Hz */
unsigned int channels; /* channels (voices) */
unsigned char reserved[16];
};
</code>
<sect2>int snd_pcm_playback_params( void *handle, snd_pcm_playback_params_t *params )
<p>
Sets various parameters for playback direction. Function returns zero if
successful, otherwise it returns an error code.
<code>
struct snd_pcm_playback_params {
int fragment_size;
int fragments_max;
int fragments_room;
unsigned char reserved[16]; /* must be filled with zero */
};
</code>
<descrip>
<tag>fragment_size</tag>
Requested size of fragment. This value should be aligned for current
format (for example to 4 if stereo 16-bit samples are used) or with the
<it>fragment_align</it> variable from <it>snd_pcm_playback_info_t</it>
structure. Its range can be from <it>min_fragment_size</it> to
<it>max_fragment_size</it>.
<tag>fragments_max</tag>
Maximum number of fragments in queue for wakeup. This number doesn't
counts partly used fragment. If current count of filled playback fragments
is greater than this value driver block application or return immediately
back if nonblock mode is active.
<tag>fragments_room</tag>
Minumum number of fragments writeable for wakeup. This value should be
in most cases 1 which means return back to application if at least
one fragment is free for playback. This value includes partly used fragments,
too.
</descrip>
<sect2>int snd_pcm_record_params( void *handle, snd_pcm_record_params_t *params )
<p>
Function sets various parameters for the recording direction. Function returns
zero if successful, otherwise it returns an error code.
<code>
struct snd_pcm_record_params {
int fragment_size;
int fragments_min;
unsigned char reserved[16];
};
</code>
<descrip>
<tag>fragment_size</tag>
Requested size of fragment. This value should be aligned for current
format (for example to 4 if stereo 16-bit samples are used) or set to the
<it>fragment_align</it> variable from <it>snd_pcm_playback_info_t</it>
structure. Its range can be from <it>min_fragment_size</it> to
<it>max_fragment_size</it>.
<tag>fragments_min</tag>
Minimum filled fragments for wakeup. Driver blocks the application (if
block mode is selected) until it isn't filled with number of fragments
specified with this value.
</descrip>
<sect2>int snd_pcm_playback_status( void *handle, snd_pcm_playback_status_t *status )
<p>
Fills the *status structure. Function returns zero if successful, otherwise
it returns an error code.
<code>
struct snd_pcm_playback_status {
unsigned int rate;
int fragments;
int fragment_size;
int count;
int queue;
int underrun;
struct timeval time;
struct timeval stime;
unsigned char reserved[16];
};
</code>
<descrip>
<tag>rate</tag>
Real playback rate. This value reflects hardware limitations.
<tag>fragments</tag>
Currently allocated fragments by the driver for playback direction.
<tag>fragment_size</tag>
Current fragment size used by driver for the playback direction.
<tag>count</tag>
Count of bytes writeable without blocking.
<tag>queue</tag>
Count of bytes in queue. Note: <it>(fragments * fragment_size) - queue</it>
should not be equal to <it>count</it>.
<tag>underrun</tag>
This value tells the application the number of underruns since the ast call
of <it>snd_pcm_playback_status</it>.
<tag>time</tag>
Delay till played of the first sample from next write. This value should
be used for time synchronization. Returned value is in the same format as
returned from the standard C function <it>gettimeofday( &ero;time, NULL )</it>.
This variable contains right value only if playback time mode is enabled
(look to <it>snd_pcm_playback_time</it> function).
<tag>stime</tag>
Time when playback was started.
This variable contains right value only if playback time mode is enabled
(look to <it>snd_pcm_playback_time</it> function).
</descrip>
<sect2>int snd_pcm_record_status( void *handle, snd_pcm_record_status_t *status )
<p>
Fills the *status structure. Function returns zero if successful, otherwise
it returns an error code.
<code>
struct snd_pcm_record_status {
unsigned int rate;
int fragments;
int fragment_size;
int count;
int free;
int overrun;
struct timeval time;
unsigned char reserved[16];
};
</code>
<descrip>
<tag>rate</tag>
Real record rate. This value reflects hardware limitations.
<tag>fragments</tag>
Currently allocated fragments by driver for the record direction.
<tag>fragment_size</tag>
Current fragment size used by driver for the record direction.
<tag>count</tag>
Count of bytes readable without blocking.
<tag>free</tag>
Count of bytes in buffer still free. Note: <it>(fragments * fragment_size) - free</it>
should not be equal to <it>count</it>.
<tag>overrun</tag>
This value tells application the count of overruns since the last call
to <it>snd_pcm_record_status</it>.
<tag>time</tag>
Lag since the next sample read was recorded. This value should be used for time
synchronization. Returned value is in the same format as returned by the
from standard C function <it>gettimeofday( &ero;time, NULL )</it>. This
variable contains right value only if record time mode is enabled (look to
<it>snd_pcm_record_time</it> function).
<tag>stime</tag>
Time when record was started. This variable contains right value only if
record time mode is enabled (look to <it>snd_pcm_record_time</it> function).
</descrip>
<sect2>int snd_pcm_drain_playback( void *handle )
<p>
This function drain playback buffers immediately. Function returns zero
if successful, otherwise it returns an error code.
<sect2>int snd_pcm_flush_playback( void *handle )
<p>
This function flushes the playback buffers. It blocks the program while the
all the waiting samples in kernel playback buffers are processed. Function
returns zero if successful, otherwise it returns an error code.
<sect2>int snd_pcm_flush_record( void *handle )
<p>
This function flushes (destroyes) record buffers. Function returns zero
if successful, otherwise it returns an error code.
<sect2>int snd_pcm_playback_time( void *handle, int enable )
<p>
This function enables or disables time mode for playback direction. Time mode
allows to application better time synchronization. Function returns zero
if successful, otherwise it returns an error code.
<sect2>int snd_pcm_record_time( void *handle, int enable )
<p>
This function enables or disables time mode for record direction. Time mode
allows to application better time synchronization. Function returns zero
if successful, otherwise it returns an error code.
<sect2>ssize_t snd_pcm_write( void *handle, const void *buffer, size_t size )
<p>
Writes samples to the device which must be in the proper format
specified by the <it>snd_pcm_playback_format</it> function. Function
returns zero or positive value if playback was successful (value represents
count of bytes which was successfuly written to device) or an
error value if error occured. Function should suspend process if
block mode is active.
<sect2>ssize_t snd_pcm_read( void *handle, void *buffer, size_t size )
<p>
Function reads samples from driver. Samples are in format specified
by <it>snd_pcm_record_format</it> function. Function returns zero
or positive value if record was success (value represents count of bytes
which was successfuly read from device) or negative error value if
error occured. Function should suspend process if block mode is active.
<sect1>Examples
<p>
The following example shows how to play the first 512kB from the
/tmp/test.au file with soundcard #0 and PCM device #0:
<tscreen><code>
int card = 0, device = 0, err, fd, count, size, idx;
void *handle;
snd_pcm_format_t format;
char *buffer;
buffer = (char *)malloc( 512 * 1024 );
if ( !buffer ) return;
if ( (err = snd_pcm_open( &ero;handle, card, device, SND_PCM_OPEN_PLAYBACK )) < 0 ) {
fprintf( stderr, "open failed: %s\n", snd_strerror( err ) );
return;
}
format.format = SND_PCM_SFMT_MU_LAW;
format.rate = 8000;
format.voices = 1;
if ( (err = snd_pcm_playback_format( handle, &ero;format )) < 0 ) {
fprintf( stderr, "format setup failed: %s\n", snd_strerror( err ) );
snd_pcm_close( handle );
return;
}
fd = open( "/tmp/test.au" );
if ( fd < 0 ) {
perror( "open file" );
snd_pcm_close( handle );
return;
}
idx = 0;
count = read( fd, buffer, 512 * 1024 );
if ( count <= 0 ) {
perror( "read from file" );
snd_pcm_close( handle );
return;
}
close( fd );
if ( !memcmp( buffer, ".snd", 4 ) ) {
idx = (buffer[4]<<24)|(buffer[5]<<16)|(buffer[6]<<8)|(buffer[7]);
if ( idx > 128 ) idx = 128;
if ( idx > count ) idx = count;
}
size = snd_pcm_write( handle, &ero;buffer[ idx ], count - idx );
printf( "Bytes written %i from %i...\n", size, count - idx );
snd_pcm_close( handle );
free( buffer );
</code></tscreen>
</article>

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