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New manual for 0.2.0 API..
This commit is contained in:
Jaroslav Kysela
parent
dc02bee60a
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104b86e839
15 changed files with 14064 additions and 3298 deletions
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EXTRA_DIST=plan.txt soundapi.txt soundapi.sgml \
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soundapi.html soundapi-1.html soundapi-2.html \
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soundapi-3.html soundapi-4.html soundapi-5.html
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EXTRA_DIST=plan.txt alsa-lib.lyx alsa-lib.html \
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2280 795 m 2160 765 l 2280 735 l 2145 735 l 2145 795 l cp
|
||||
clip
|
||||
n 2160 765 m 6615 765 l gs col0 s gr gr
|
||||
|
||||
% arrowhead
|
||||
n 2280 795 m 2160 765 l 2280 735 l col0 s
|
||||
% arrowhead
|
||||
n 6495 735 m 6615 765 l 6495 795 l col0 s
|
||||
% Polyline
|
||||
gs clippath
|
||||
6915 795 m 6795 765 l 6915 735 l 6780 735 l 6780 795 l cp
|
||||
clip
|
||||
n 6795 765 m 8415 765 l gs col0 s gr gr
|
||||
|
||||
% arrowhead
|
||||
n 6915 795 m 6795 765 l 6915 735 l col0 s
|
||||
% Polyline
|
||||
[45 22 15 22] 0 sd
|
||||
n 1980 900 m 1980 1395 l gs col0 s gr [] 0 sd
|
||||
% Polyline
|
||||
[15 45] 45 sd
|
||||
n 6795 900 m 6795 1395 l gs col0 s gr [] 0 sd
|
||||
% Polyline
|
||||
n 1215 900 m 1665 1395 l 1980 900 l gs col0 s gr
|
||||
% Polyline
|
||||
n 1215 1395 m 1665 900 l 1980 1395 l gs col0 s gr
|
||||
/Times-Roman ff 180.00 scf sf
|
||||
1170 1615 m
|
||||
gs 1 -1 sc (0) col0 sh gr
|
||||
/Times-Roman ff 180.00 scf sf
|
||||
2970 1615 m
|
||||
gs 1 -1 sc (4) col0 sh gr
|
||||
/Times-Roman ff 180.00 scf sf
|
||||
4770 1615 m
|
||||
gs 1 -1 sc (8) col0 sh gr
|
||||
/Times-Roman ff 180.00 scf sf
|
||||
6525 1615 m
|
||||
gs 1 -1 sc (12) col0 sh gr
|
||||
/Times-Roman ff 180.00 scf sf
|
||||
8325 1615 m
|
||||
gs 1 -1 sc (16) col0 sh gr
|
||||
/Times-Roman ff 180.00 scf sf
|
||||
7515 630 m
|
||||
gs 1 -1 sc (count) col0 sh gr
|
||||
/Times-Roman ff 180.00 scf sf
|
||||
1350 630 m
|
||||
gs 1 -1 sc (count) col0 sh gr
|
||||
/Times-Roman ff 180.00 scf sf
|
||||
4095 630 m
|
||||
gs 1 -1 sc (free) col0 sh gr
|
||||
$F2psEnd
|
||||
rs
|
||||
showpage
|
||||
|
|
@ -1,34 +0,0 @@
|
|||
<HTML>
|
||||
<HEAD>
|
||||
<TITLE>Advanced Linux Sound Architecture - Library API: Introduction</TITLE>
|
||||
</HEAD>
|
||||
<BODY>
|
||||
Previous
|
||||
<A HREF="soundapi-2.html">Next</A>
|
||||
<A HREF="soundapi.html#toc1">Table of Contents</A>
|
||||
<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/asoundlib.h</LI>
|
||||
<LI>/usr/include/linux/asound.h</LI>
|
||||
<LI>/usr/include/linux/asoundid.h</LI>
|
||||
</OL>
|
||||
</P>
|
||||
|
||||
<HR>
|
||||
Previous
|
||||
<A HREF="soundapi-2.html">Next</A>
|
||||
<A HREF="soundapi.html#toc1">Table of Contents</A>
|
||||
</BODY>
|
||||
</HTML>
|
||||
|
|
@ -1,44 +0,0 @@
|
|||
<HTML>
|
||||
<HEAD>
|
||||
<TITLE>Advanced Linux Sound Architecture - Library API: Error Codes</TITLE>
|
||||
</HEAD>
|
||||
<BODY>
|
||||
<A HREF="soundapi-1.html">Previous</A>
|
||||
<A HREF="soundapi-3.html">Next</A>
|
||||
<A HREF="soundapi.html#toc2">Table of Contents</A>
|
||||
<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>
|
||||
<A HREF="soundapi-1.html">Previous</A>
|
||||
<A HREF="soundapi-3.html">Next</A>
|
||||
<A HREF="soundapi.html#toc2">Table of Contents</A>
|
||||
</BODY>
|
||||
</HTML>
|
||||
|
|
@ -1,159 +0,0 @@
|
|||
<HTML>
|
||||
<HEAD>
|
||||
<TITLE>Advanced Linux Sound Architecture - Library API: Control Interface</TITLE>
|
||||
</HEAD>
|
||||
<BODY>
|
||||
<A HREF="soundapi-2.html">Previous</A>
|
||||
<A HREF="soundapi-4.html">Next</A>
|
||||
<A HREF="soundapi.html#toc3">Table of Contents</A>
|
||||
<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( &handle, card )) < 0 ) {
|
||||
fprintf( stderr, "open failed: %s\n", snd_strerror( err ) );
|
||||
return;
|
||||
}
|
||||
if ( (err = snd_ctl_hw_info( handle, &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 );
|
||||
</PRE>
|
||||
<HR>
|
||||
</CODE></BLOCKQUOTE>
|
||||
</P>
|
||||
|
||||
|
||||
<HR>
|
||||
<A HREF="soundapi-2.html">Previous</A>
|
||||
<A HREF="soundapi-4.html">Next</A>
|
||||
<A HREF="soundapi.html#toc3">Table of Contents</A>
|
||||
</BODY>
|
||||
</HTML>
|
||||
|
|
@ -1,268 +0,0 @@
|
|||
<HTML>
|
||||
<HEAD>
|
||||
<TITLE>Advanced Linux Sound Architecture - Library API: Mixer Interface</TITLE>
|
||||
</HEAD>
|
||||
<BODY>
|
||||
<A HREF="soundapi-3.html">Previous</A>
|
||||
<A HREF="soundapi-5.html">Next</A>
|
||||
<A HREF="soundapi.html#toc4">Table of Contents</A>
|
||||
<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 "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"
|
||||
|
||||
</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( &handle, card, device )) < 0 ) {
|
||||
fprintf( stderr, "open failed: %s\n", snd_strerror( err ) );
|
||||
return;
|
||||
}
|
||||
if ( (err = snd_mixer_info( handle, &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, &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, &channel )) < 0 ) {
|
||||
fprintf( stderr, "master write failed: %s\n", snd_strerror( err ) );
|
||||
snd_mixer_close( handle );
|
||||
return;
|
||||
}
|
||||
}
|
||||
snd_mixer_close( handle );
|
||||
</PRE>
|
||||
<HR>
|
||||
</CODE></BLOCKQUOTE>
|
||||
</P>
|
||||
|
||||
|
||||
<HR>
|
||||
<A HREF="soundapi-3.html">Previous</A>
|
||||
<A HREF="soundapi-5.html">Next</A>
|
||||
<A HREF="soundapi.html#toc4">Table of Contents</A>
|
||||
</BODY>
|
||||
</HTML>
|
||||
|
|
@ -1,552 +0,0 @@
|
|||
<HTML>
|
||||
<HEAD>
|
||||
<TITLE>Advanced Linux Sound Architecture - Library API: Digital Audio (PCM) Interface</TITLE>
|
||||
</HEAD>
|
||||
<BODY>
|
||||
<A HREF="soundapi-4.html">Previous</A>
|
||||
Next
|
||||
<A HREF="soundapi.html#toc5">Table of Contents</A>
|
||||
<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 << 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)
|
||||
</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( &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( &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( &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.channels = 1;
|
||||
if ( (err = snd_pcm_playback_format( handle, &format )) < 0 ) {
|
||||
fprintf( stderr, "format setup failed: %s\n", snd_strerror( err ) );
|
||||
snd_pcm_close( handle );
|
||||
return;
|
||||
}
|
||||
fd = open( "/tmp/test.au", O_RDONLY );
|
||||
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, &buffer[ idx ], count - idx );
|
||||
printf( "Bytes written %i from %i...\n", size, count - idx );
|
||||
snd_pcm_close( handle );
|
||||
free( buffer );
|
||||
</PRE>
|
||||
<HR>
|
||||
</CODE></BLOCKQUOTE>
|
||||
</P>
|
||||
|
||||
|
||||
<HR>
|
||||
<A HREF="soundapi-4.html">Previous</A>
|
||||
Next
|
||||
<A HREF="soundapi.html#toc5">Table of Contents</A>
|
||||
</BODY>
|
||||
</HTML>
|
||||
|
|
@ -1,52 +0,0 @@
|
|||
|
||||
<HTML>
|
||||
<HEAD>
|
||||
<TITLE>Advanced Linux Sound Architecture - Library API</TITLE>
|
||||
</HEAD>
|
||||
<BODY>
|
||||
Previous
|
||||
<A HREF="soundapi-1.html">Next</A>
|
||||
Table of Contents
|
||||
<HR>
|
||||
<H1>Advanced Linux Sound Architecture - Library API</H1>
|
||||
|
||||
<H2>Jaroslav Kysela <CODE><perex@jcu.cz></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>
|
||||
</UL>
|
||||
|
||||
|
||||
<HR>
|
||||
Previous
|
||||
<A HREF="soundapi-1.html">Next</A>
|
||||
Table of Contents
|
||||
</BODY>
|
||||
</HTML>
|
||||
|
|
@ -1,956 +0,0 @@
|
|||
<!doctype linuxdoc system>
|
||||
|
||||
<!-- Advanced Linux Sound Architecture - Library API -->
|
||||
|
||||
<article>
|
||||
|
||||
<!-- Title information -->
|
||||
|
||||
<title>Advanced Linux Sound Architecture - Library API
|
||||
<author>Jaroslav Kysela <tt><perex@jcu.cz></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/asoundlib.h
|
||||
<item>/usr/include/linux/asound.h
|
||||
<item>/usr/include/linux/asoundid.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.channels = 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", O_RDONLY );
|
||||
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>
|
||||
1230
doc/soundapi.txt
1230
doc/soundapi.txt
File diff suppressed because it is too large
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