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Jaroslav Kysela 1998-08-13 15:42:56 +00:00
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41
CHANGELOG Normal file
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0.0.8 -> 0.0.9
* Makefile and configure.in changes
- added check for alsa driver package
- added spec file for RPM
0.0.7 -> 0.0.8
* added LGPL notice to all source and header files
0.0.6 -> 0.0.7
* added snd_cards_name function
0.0.5 -> 0.0.6
* fixed SND_PCM_OPEN constants
0.0.4 -> 0.0.5
* added snd_cards_mask function
* added info functions for pcm playback/record in control interface
* fixed Makefile bugs for shared library (added -fPIC)
0.0.3 -> 0.0.4
* changed COPYING policy from GPL to LGPL
* fixed bug in snd_mixer_channel_read & write
* added mixer exact support
* added pcm time mode support
* 'make install' is now possible
0.0.2 -> 0.0.3
* corrected documentation
0.0.1 -> 0.0.2
* added file COPYING
* added documentation in sgml + plan.txt
* minor changes in API for MIXER & PCM

481
COPYING Normal file
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GNU LIBRARY GENERAL PUBLIC LICENSE
Version 2, June 1991
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675 Mass Ave, Cambridge, MA 02139, USA
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#
# Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
ifeq (Makefile.conf,$(wildcard Makefile.conf))
include Makefile.conf
else
dummy:
@echo
@echo "Please, run configure script as first..."
@echo
endif
all: include/soundlib.h
$(MAKE) -C src
$(MAKE) -C doc
@echo
@echo "ALSA library were sucessfully compiled."
@echo
include/soundlib.h: include/header.h include/version.h include/error.h include/footer.h \
include/control.h include/mixer.h include/pcm.h
cat include/header.h include/version.h include/error.h \
include/control.h include/mixer.h \
include/pcm.h \
include/footer.h > include/soundlib.h
install: all
$(INSTALL) -m 644 -o root -g root include/soundlib.h ${includedir}/sys
$(INSTALL) -m 644 -o root -g root lib/libsound.a ${libdir}
$(LN_S) -f libsound.so.${SND_LIB_VERSION} ${libdir}/libsound.so
$(LN_S) -f libsound.so.${SND_LIB_VERSION} ${libdir}/libsound.so.${SND_LIB_MAJOR}
$(INSTALL) -m 644 -o root -g root lib/libsound.so.${SND_LIB_VERSION} ${libdir}
/sbin/ldconfig
clean:
$(MAKE) -C include clean
$(MAKE) -C src clean
$(MAKE) -C test clean
$(MAKE) -C doc clean
$(MAKE) -C utils clean
rm -f core .depend *.o *.orig *~
rm -f `find . -name "out.txt"`
pack: clean
rm -f config.cache config.log config.status Makefile.conf
chown -R root.root ../alsa-lib
tar cvz -C .. -f ../alsa-lib-$(SND_LIB_VERSION).tar.gz alsa-lib
publish: pack
cat ../alsa-lib-$(SND_LIB_VERSION).tar.gz | \
ssh -l root zarquon.jcu.cz /home/alsa/publishlib alsa-lib-$(SND_LIB_VERSION).tar.gz

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#
# Configuration Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
srcdir=@srcdir@
VPATH=@srcdir@
prefix=@prefix@
exec_prefix=@exec_prefix@
includedir=@includedir@
libdir=@libdir@
c_opts=@CFLAGS@
INSTALL=@INSTALL@
SND_LIB_VERSION=@SND_LIB_VERSION@
SND_LIB_MAJOR=@SND_LIB_MAJOR@
SND_LIB_MINOR=@SND_LIB_MINOR@
SND_LIB_SUBMINOR=@SND_LIB_SUBMINOR@
CC=@CC@
CPP=@CPP@
INCLUDE=-I../include -I../../include
COPTS = $(c_opts)
COPTS += -Wall -Wstrict-prototypes -fomit-frame-pointer -pipe
LINKER=ld
LN_S=@LN_S@

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dnl Process this file with autoconf to produce a configure script.
AC_INIT(Makefile.conf.in)
AC_PREFIX_DEFAULT(/usr)
dnl Checks for programs.
AC_PROG_CC
AC_PROG_RANLIB
AC_PROG_INSTALL
AC_PROG_LN_S
dnl Checks for header files.
AC_HEADER_STDC
AC_CONFIG_HEADER(include/config.h)
AC_CHECK_HEADERS(linux/sound.h)
dnl Checks for typedefs, structures, and compiler characteristics.
AC_C_CONST
AC_C_INLINE
AC_C_BIGENDIAN
AC_HEADER_TIME
dnl Checks for library functions.
AC_PROG_GCC_TRADITIONAL
dnl Check for ALSA driver package.
myprefix=$prefix
if test "$myprefix" = "NONE"; then
myprefix=$ac_default_prefix
fi
CFLAGS="-I$myprefix/include"
#echo "CFLAGS=$CFLAGS"
AC_MSG_CHECKING(for alsa-driver package)
AC_TRY_RUN([
#include <linux/sound.h>
void main(void)
{
#ifndef SND_PROTOCOL_VERSION
exit(1);
#else
exit(0);
#endif
}
],
AC_MSG_RESULT("present"),
AC_MSG_RESULT("not found"); echo "Fatal error: Install alsa-driver package at first..."; exit 1;,
AC_MSG_RESULT("not supported"); echo "Fatal error: Cross-compiling isn't supported..."; exit 1;,
)
dnl Check for version...
AC_MSG_CHECKING(for library version)
SND_LIB_VERSION=`cat $srcdir/version`
AC_DEFINE_UNQUOTED(SND_LIB_VERSION, "$SND_LIB_VERSION")
AC_SUBST(SND_LIB_VERSION)
SND_LIB_MAJOR=`echo $SND_LIB_VERSION | cut -d . -f 1`
AC_SUBST(SND_LIB_MAJOR)
SND_LIB_MINOR=`echo $SND_LIB_VERSION | cut -d . -f 2`
AC_SUBST(SND_LIB_MINOR)
SND_LIB_SUBMINOR=`echo $SND_LIB_VERSION | cut -d . -f 3`
AC_SUBST(SND_LIB_SUBMINOR)
AC_MSG_RESULT($SND_LIB_VERSION)
AC_OUTPUT(Makefile.conf include/version.h utils/alsa-lib.spec)

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

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

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

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#
# Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
include ../Makefile.conf
clean:
rm -f core .depend *.o *.orig *~

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/* include/config.h. Generated automatically by configure. */
/*
* Configuration header file for compilation of the ALSA driver
*/
#define SND_LIB_VERSION "0.0.9"
/* #undef WORDS_BIGENDIAN */

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/*
* Configuration header file for compilation of the ALSA driver
*/
#undef SND_LIB_VERSION
#undef WORDS_BIGENDIAN

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/****************************************************************************
* *
* control.h *
* Control Interface *
* *
****************************************************************************/
#ifdef __cplusplus
extern "C" {
#endif
int snd_cards( void );
unsigned int snd_cards_mask( void );
int snd_card_name( const char *name );
int snd_ctl_open( void **handle, int card );
int snd_ctl_close( void *handle );
int snd_ctl_file_descriptor( void *handle );
int snd_ctl_hw_info( void *handle, struct snd_ctl_hw_info *info );
int snd_ctl_pcm_info( void *handle, int dev, snd_pcm_info_t *info );
int snd_ctl_pcm_playback_info( void *handle, int dev, snd_pcm_playback_info_t *info );
int snd_ctl_pcm_record_info( void *handle, int dev, snd_pcm_record_info_t *info );
int snd_ctl_mixer_info( void *handle, int dev, snd_mixer_info_t *info );
#ifdef __cplusplus
}
#endif

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/*
* error.h
*/
#define SND_ERROR_BEGIN 500000
#define SND_ERROR_UNCOMPATIBLE_VERSION (SND_ERROR_BEGIN+0)
#ifdef __cplusplus
extern "C" {
#endif
const char *snd_strerror( int errnum );
#ifdef __cplusplus
}
#endif

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/*
*
*/
#endif /* __SOUNDLIB_H */

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/*
* Application interface library for the ALSA driver
* Copyright (c) by Jaroslav Kysela <perex@jcu.cz>
*
*
* This library is free software; you can redistribute it and/or modify
* it under the terms of the GNU Library General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#ifndef __SOUNDLIB_H
#define __SOUNDLIB_H
#include <linux/sound.h>
#include <unistd.h>

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/*
* Application interface library for the ALSA driver
* Copyright (c) 1994/98 by Jaroslav Kysela <perex@jcu.cz>
*/
#include <asm/byteorder.h>
#include "../../../include/ultraconf.h"
#ifdef DEBUG
#include <assert.h>
#define ASSERT( s ) assert( s )
void __snd_dprintf( char *, ... );
#define snd_dprintf( args... ) __ultra_dprintf( ##args )
#else
#define ASSERT( s ) { ; }
#define snd_dprintf( fmt... ) /* nothing */
#endif
extern inline int snd_p_rint( float x )
{
return (int)( x + (float)0.5 );
}
/*
*
*/
extern inline void snd_put_byte( unsigned char *array, unsigned int idx, unsigned char b )
{
*(array + idx) = b;
}
extern inline unsigned char snd_get_byte( unsigned char *array, unsigned int idx )
{
return *(array + idx);
}
#if defined( __i386__ )
extern inline void snd_put_le_word( unsigned char *array, unsigned int idx, unsigned short w )
{
*(unsigned short *)(array + idx) = w;
}
extern inline unsigned short snd_get_le_word( unsigned char *array, unsigned int idx )
{
return *(unsigned short *)(array + idx);
}
extern inline void snd_put_le_dword( unsigned char *array, unsigned int idx, unsigned int dw )
{
*(unsigned int *)(array + idx) = dw;
}
extern inline unsigned int snd_get_le_dword( unsigned char *array, unsigned int idx )
{
return *(unsigned int *)(array + idx );
}
#else
#ifndef WORDS_BIGENDIAN
extern inline void snd_put_le_word( unsigned char *array, unsigned int idx, unsigned short w )
{
*(array + idx + 0) = (unsigned char)(w >> 0);
*(array + idx + 1) = (unsigned char)(w >> 8);
}
extern inline unsigned short snd_get_le_word( unsigned char *array, unsigned int idx )
{
return ( *(array + idx + 0) << 0 ) |
( *(array + idx + 1) << 8 );
}
extern inline void snd_put_le_dword( unsigned char *array, unsigned int idx, unsigned int dw )
{
*(array + idx + 0) = (unsigned char)(dw >> 0);
*(array + idx + 1) = (unsigned char)(dw >> 8);
*(array + idx + 2) = (unsigned char)(dw >> 16);
*(array + idx + 3) = (unsigned char)(dw >> 24);
}
extern inline unsigned int snd_get_le_dword( unsigned char *array, unsigned int idx )
{
return ( *(array + idx + 0) << 0 ) |
( *(array + idx + 1) << 8 ) |
( *(array + idx + 2) << 16 ) |
( *(array + idx + 3) << 24 );
}
#else
extern inline void snd_put_le_word( unsigned char *array, unsigned int idx, unsigned short w )
{
*(array + idx + 0) = (unsigned char)(w >> 8);
*(array + idx + 1) = (unsigned char)(w >> 0);
}
extern inline unsigned short snd_get_le_word( unsigned char *array, unsigned int idx )
{
return ( *(array + idx + 0) << 8 ) |
( *(array + idx + 1) << 0 );
}
extern inline void snd_put_le_dword( unsigned char *array, unsigned int idx, unsigned int dw )
{
*(array + idx + 0) = (unsigned char)(dw >> 24);
*(array + idx + 1) = (unsigned char)(dw >> 16);
*(array + idx + 2) = (unsigned char)(dw >> 8);
*(array + idx + 3) = (unsigned char)(dw >> 0);
}
extern inline unsigned int snd_get_le_dword( unsigned char *array, unsigned int idx )
{
return ( *(array + idx + 0) << 24 ) |
( *(array + idx + 1) << 16 ) |
( *(array + idx + 2) << 8 ) |
( *(array + idx + 3) << 0 );
}
#endif
#endif
/*
*
*/
extern inline unsigned char snd_get_unsigned_byte( unsigned char *ptr )
{
return *ptr;
}
extern inline void snd_put_unsigned_byte( unsigned char *ptr, unsigned char val )
{
*ptr = val;
}
extern inline unsigned char snd_get_signed_byte( signed char *ptr )
{
return *ptr;
}
extern inline void snd_put_signed_byte( signed char *ptr, signed char val )
{
*ptr = val;
}
#if !defined( WORDS_BIGENDIAN ) || defined( __i386__ )
extern inline signed short snd_get_signed_le_word( unsigned char *ptr )
{
return *(signed short *)ptr;
}
extern inline void snd_put_signed_le_word( unsigned char *ptr, signed short val )
{
*(signed short *)ptr = val;
}
extern inline unsigned short snd_get_unsigned_le_word( unsigned char *ptr )
{
return *(unsigned short *)ptr;
}
extern inline void snd_put_unsigned_le_word( unsigned char *ptr, unsigned short val )
{
*(unsigned short *)ptr = val;
}
extern inline unsigned int snd_get_unsigned_le_dword( unsigned char *ptr )
{
return *(unsigned int *)ptr;
}
extern inline void snd_put_unsigned_le_dword( unsigned char *ptr, unsigned int val )
{
*(unsigned int *)ptr = val;
}
#else
extern inline unsigned short snd_get_signed_le_word( unsigned char *ptr )
{
return (signed short)( *ptr + ( *( ptr + 1 ) << 8 );
}
extern inline void snd_put_signed_le_word( unsigned char *ptr, signed short val )
{
*ptr = (unsigned char)val;
*(ptr + 1) = val >> 8;
}
extern inline unsigned short snd_get_unsigned_le_word( unsigned char *ptr )
{
return (signed short)( *ptr + ( *( ptr + 1 ) << 8 );
}
extern inline void snd_put_unsigned_le_word( unsigned char *ptr, unsigned short val )
{
*ptr = (unsigned char)val;
*(ptr + 1) = val >> 8;
}
extern inline unsigned int snd_get_unsigned_le_dword( unsigned char *ptr )
{
return (signed int)( *ptr + ( *( ptr + 1 ) << 8 ) +
( *( ptr + 2 ) << 8 ) + ( *( ptr + 3 ) << 8 ) );
}
extern inline void snd_put_unsigned_le_dword( unsigned char *ptr, unsigned int val )
{
*ptr = (unsigned char)val;
*(ptr + 1) = (unsigned char)val >> 8;
*(ptr + 2) = (unsigned char)val >> 16;
*(ptr + 3) = (unsigned char)val >> 24;
}
#endif
extern inline unsigned short snd_get_be_word( unsigned char *array, unsigned int idx )
{
return ( *(array + idx + 1) << 0 ) |
( *(array + idx + 0) << 8 );
}
extern inline unsigned int snd_get_be_dword( unsigned char *array, unsigned int idx )
{
return ( *(array + idx + 3) << 0 ) |
( *(array + idx + 2) << 8 ) |
( *(array + idx + 1) << 16 ) |
( *(array + idx + 0) << 24 );
}

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/****************************************************************************
* *
* mixer.h *
* Mixer Interface *
* *
****************************************************************************/
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;
#ifdef __cplusplus
extern "C" {
#endif
int snd_mixer_open( void **handle, int card, int device );
int snd_mixer_close( void *handle );
int snd_mixer_file_descriptor( void *handle );
int snd_mixer_channels( void *handle );
int snd_mixer_info( void *handle, snd_mixer_info_t *info );
int snd_mixer_exact_mode( void *handle, int enable );
int snd_mixer_channel( void *handle, const char *channel_id );
int snd_mixer_channel_info( void *handle, int channel, snd_mixer_channel_info_t *info );
int snd_mixer_channel_read( void *handle, int channel, snd_mixer_channel_t *data );
int snd_mixer_channel_write( void *handle, int channel, snd_mixer_channel_t *data );
int snd_mixer_special_read( void *handle, snd_mixer_special_t *special );
int snd_mixer_special_write( void *handle, snd_mixer_special_t *special );
int snd_mixer_read( void *handle, snd_mixer_callbacks_t *callbacks );
#ifdef __cplusplus
}
#endif

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/****************************************************************************
* *
* pcm.h *
* Digital Audio Interface *
* *
****************************************************************************/
#define SND_PCM_OPEN_PLAYBACK (O_WRONLY)
#define SND_PCM_OPEN_RECORD (O_RDONLY)
#define SND_PCM_OPEN_DUPLEX (O_RDWR)
#ifdef __cplusplus
extern "C" {
#endif
int snd_pcm_open( void **handle, int card, int device, int mode );
int snd_pcm_close( void *handle );
int snd_pcm_file_descriptor( void *handle );
int snd_pcm_block_mode( void *handle, int enable );
int snd_pcm_info( void *handle, snd_pcm_info_t *info );
int snd_pcm_playback_info( void *handle, snd_pcm_playback_info_t *info );
int snd_pcm_record_info( void *handle, snd_pcm_record_info_t *info );
int snd_pcm_playback_format( void *handle, snd_pcm_format_t *format );
int snd_pcm_record_format( void *handle, snd_pcm_format_t *format );
int snd_pcm_playback_params( void *handle, snd_pcm_playback_params_t *params );
int snd_pcm_record_params( void *handle, snd_pcm_record_params_t *params );
int snd_pcm_playback_status( void *handle, snd_pcm_playback_status_t *status );
int snd_pcm_record_status( void *handle, snd_pcm_record_status_t *status );
int snd_pcm_drain_playback( void *handle );
int snd_pcm_flush_playback( void *handle );
int snd_pcm_flush_record( void *handle );
int snd_pcm_playback_time( void *handle, int enable );
int snd_pcm_record_time( void *handle, int enable );
ssize_t snd_pcm_write( void *handle, const void *buffer, size_t size );
ssize_t snd_pcm_read( void *handle, void *buffer, size_t size );
#ifdef __cplusplus
}
#endif

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/*
* Application interface library for the ALSA driver
* Copyright (c) by Jaroslav Kysela <perex@jcu.cz>
*
*
* This library is free software; you can redistribute it and/or modify
* it under the terms of the GNU Library General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#ifndef __SOUNDLIB_H
#define __SOUNDLIB_H
#include <linux/sound.h>
#include <unistd.h>
/*
* version.h
*/
#define SOUNDLIB_VERSION_MAJOR 0
#define SOUNDLIB_VERSION_MINOR 0
#define SOUNDLIB_VERSION_SUBMINOR 9
#define SOUNDLIB_VERSION ( ( LIBULTRA_VERSION_MAJOR << 16 ) | ( LIBULTRA_VERSION_MINOR << 8 ) | LIB_ULTRA_VERSION_SUBMINOR )
/*
* error.h
*/
#define SND_ERROR_BEGIN 500000
#define SND_ERROR_UNCOMPATIBLE_VERSION (SND_ERROR_BEGIN+0)
#ifdef __cplusplus
extern "C" {
#endif
const char *snd_strerror( int errnum );
#ifdef __cplusplus
}
#endif
/****************************************************************************
* *
* control.h *
* Control Interface *
* *
****************************************************************************/
#ifdef __cplusplus
extern "C" {
#endif
int snd_cards( void );
unsigned int snd_cards_mask( void );
int snd_card_name( const char *name );
int snd_ctl_open( void **handle, int card );
int snd_ctl_close( void *handle );
int snd_ctl_file_descriptor( void *handle );
int snd_ctl_hw_info( void *handle, struct snd_ctl_hw_info *info );
int snd_ctl_pcm_info( void *handle, int dev, snd_pcm_info_t *info );
int snd_ctl_pcm_playback_info( void *handle, int dev, snd_pcm_playback_info_t *info );
int snd_ctl_pcm_record_info( void *handle, int dev, snd_pcm_record_info_t *info );
int snd_ctl_mixer_info( void *handle, int dev, snd_mixer_info_t *info );
#ifdef __cplusplus
}
#endif
/****************************************************************************
* *
* mixer.h *
* Mixer Interface *
* *
****************************************************************************/
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;
#ifdef __cplusplus
extern "C" {
#endif
int snd_mixer_open( void **handle, int card, int device );
int snd_mixer_close( void *handle );
int snd_mixer_file_descriptor( void *handle );
int snd_mixer_channels( void *handle );
int snd_mixer_info( void *handle, snd_mixer_info_t *info );
int snd_mixer_exact_mode( void *handle, int enable );
int snd_mixer_channel( void *handle, const char *channel_id );
int snd_mixer_channel_info( void *handle, int channel, snd_mixer_channel_info_t *info );
int snd_mixer_channel_read( void *handle, int channel, snd_mixer_channel_t *data );
int snd_mixer_channel_write( void *handle, int channel, snd_mixer_channel_t *data );
int snd_mixer_special_read( void *handle, snd_mixer_special_t *special );
int snd_mixer_special_write( void *handle, snd_mixer_special_t *special );
int snd_mixer_read( void *handle, snd_mixer_callbacks_t *callbacks );
#ifdef __cplusplus
}
#endif
/****************************************************************************
* *
* pcm.h *
* Digital Audio Interface *
* *
****************************************************************************/
#define SND_PCM_OPEN_PLAYBACK (O_WRONLY)
#define SND_PCM_OPEN_RECORD (O_RDONLY)
#define SND_PCM_OPEN_DUPLEX (O_RDWR)
#ifdef __cplusplus
extern "C" {
#endif
int snd_pcm_open( void **handle, int card, int device, int mode );
int snd_pcm_close( void *handle );
int snd_pcm_file_descriptor( void *handle );
int snd_pcm_block_mode( void *handle, int enable );
int snd_pcm_info( void *handle, snd_pcm_info_t *info );
int snd_pcm_playback_info( void *handle, snd_pcm_playback_info_t *info );
int snd_pcm_record_info( void *handle, snd_pcm_record_info_t *info );
int snd_pcm_playback_format( void *handle, snd_pcm_format_t *format );
int snd_pcm_record_format( void *handle, snd_pcm_format_t *format );
int snd_pcm_playback_params( void *handle, snd_pcm_playback_params_t *params );
int snd_pcm_record_params( void *handle, snd_pcm_record_params_t *params );
int snd_pcm_playback_status( void *handle, snd_pcm_playback_status_t *status );
int snd_pcm_record_status( void *handle, snd_pcm_record_status_t *status );
int snd_pcm_drain_playback( void *handle );
int snd_pcm_flush_playback( void *handle );
int snd_pcm_flush_record( void *handle );
int snd_pcm_playback_time( void *handle, int enable );
int snd_pcm_record_time( void *handle, int enable );
ssize_t snd_pcm_write( void *handle, const void *buffer, size_t size );
ssize_t snd_pcm_read( void *handle, void *buffer, size_t size );
#ifdef __cplusplus
}
#endif
/*
*
*/
#endif /* __SOUNDLIB_H */

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/*
* version.h
*/
#define SOUNDLIB_VERSION_MAJOR 0
#define SOUNDLIB_VERSION_MINOR 0
#define SOUNDLIB_VERSION_SUBMINOR 9
#define SOUNDLIB_VERSION ( ( LIBULTRA_VERSION_MAJOR << 16 ) | ( LIBULTRA_VERSION_MINOR << 8 ) | LIB_ULTRA_VERSION_SUBMINOR )

9
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@ -0,0 +1,9 @@
/*
* version.h
*/
#define SOUNDLIB_VERSION_MAJOR @SND_LIB_MAJOR@
#define SOUNDLIB_VERSION_MINOR @SND_LIB_MINOR@
#define SOUNDLIB_VERSION_SUBMINOR @SND_LIB_SUBMINOR@
#define SOUNDLIB_VERSION ( ( LIBULTRA_VERSION_MAJOR << 16 ) | ( LIBULTRA_VERSION_MINOR << 8 ) | LIB_ULTRA_VERSION_SUBMINOR )

250
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#! /bin/sh
#
# install - install a program, script, or datafile
# This comes from X11R5 (mit/util/scripts/install.sh).
#
# Copyright 1991 by the Massachusetts Institute of Technology
#
# Permission to use, copy, modify, distribute, and sell this software and its
# documentation for any purpose is hereby granted without fee, provided that
# the above copyright notice appear in all copies and that both that
# copyright notice and this permission notice appear in supporting
# documentation, and that the name of M.I.T. not be used in advertising or
# publicity pertaining to distribution of the software without specific,
# written prior permission. M.I.T. makes no representations about the
# suitability of this software for any purpose. It is provided "as is"
# without express or implied warranty.
#
# Calling this script install-sh is preferred over install.sh, to prevent
# `make' implicit rules from creating a file called install from it
# when there is no Makefile.
#
# This script is compatible with the BSD install script, but was written
# from scratch. It can only install one file at a time, a restriction
# shared with many OS's install programs.
# set DOITPROG to echo to test this script
# Don't use :- since 4.3BSD and earlier shells don't like it.
doit="${DOITPROG-}"
# put in absolute paths if you don't have them in your path; or use env. vars.
mvprog="${MVPROG-mv}"
cpprog="${CPPROG-cp}"
chmodprog="${CHMODPROG-chmod}"
chownprog="${CHOWNPROG-chown}"
chgrpprog="${CHGRPPROG-chgrp}"
stripprog="${STRIPPROG-strip}"
rmprog="${RMPROG-rm}"
mkdirprog="${MKDIRPROG-mkdir}"
transformbasename=""
transform_arg=""
instcmd="$mvprog"
chmodcmd="$chmodprog 0755"
chowncmd=""
chgrpcmd=""
stripcmd=""
rmcmd="$rmprog -f"
mvcmd="$mvprog"
src=""
dst=""
dir_arg=""
while [ x"$1" != x ]; do
case $1 in
-c) instcmd="$cpprog"
shift
continue;;
-d) dir_arg=true
shift
continue;;
-m) chmodcmd="$chmodprog $2"
shift
shift
continue;;
-o) chowncmd="$chownprog $2"
shift
shift
continue;;
-g) chgrpcmd="$chgrpprog $2"
shift
shift
continue;;
-s) stripcmd="$stripprog"
shift
continue;;
-t=*) transformarg=`echo $1 | sed 's/-t=//'`
shift
continue;;
-b=*) transformbasename=`echo $1 | sed 's/-b=//'`
shift
continue;;
*) if [ x"$src" = x ]
then
src=$1
else
# this colon is to work around a 386BSD /bin/sh bug
:
dst=$1
fi
shift
continue;;
esac
done
if [ x"$src" = x ]
then
echo "install: no input file specified"
exit 1
else
true
fi
if [ x"$dir_arg" != x ]; then
dst=$src
src=""
if [ -d $dst ]; then
instcmd=:
else
instcmd=mkdir
fi
else
# Waiting for this to be detected by the "$instcmd $src $dsttmp" command
# might cause directories to be created, which would be especially bad
# if $src (and thus $dsttmp) contains '*'.
if [ -f $src -o -d $src ]
then
true
else
echo "install: $src does not exist"
exit 1
fi
if [ x"$dst" = x ]
then
echo "install: no destination specified"
exit 1
else
true
fi
# If destination is a directory, append the input filename; if your system
# does not like double slashes in filenames, you may need to add some logic
if [ -d $dst ]
then
dst="$dst"/`basename $src`
else
true
fi
fi
## this sed command emulates the dirname command
dstdir=`echo $dst | sed -e 's,[^/]*$,,;s,/$,,;s,^$,.,'`
# Make sure that the destination directory exists.
# this part is taken from Noah Friedman's mkinstalldirs script
# Skip lots of stat calls in the usual case.
if [ ! -d "$dstdir" ]; then
defaultIFS='
'
IFS="${IFS-${defaultIFS}}"
oIFS="${IFS}"
# Some sh's can't handle IFS=/ for some reason.
IFS='%'
set - `echo ${dstdir} | sed -e 's@/@%@g' -e 's@^%@/@'`
IFS="${oIFS}"
pathcomp=''
while [ $# -ne 0 ] ; do
pathcomp="${pathcomp}${1}"
shift
if [ ! -d "${pathcomp}" ] ;
then
$mkdirprog "${pathcomp}"
else
true
fi
pathcomp="${pathcomp}/"
done
fi
if [ x"$dir_arg" != x ]
then
$doit $instcmd $dst &&
if [ x"$chowncmd" != x ]; then $doit $chowncmd $dst; else true ; fi &&
if [ x"$chgrpcmd" != x ]; then $doit $chgrpcmd $dst; else true ; fi &&
if [ x"$stripcmd" != x ]; then $doit $stripcmd $dst; else true ; fi &&
if [ x"$chmodcmd" != x ]; then $doit $chmodcmd $dst; else true ; fi
else
# If we're going to rename the final executable, determine the name now.
if [ x"$transformarg" = x ]
then
dstfile=`basename $dst`
else
dstfile=`basename $dst $transformbasename |
sed $transformarg`$transformbasename
fi
# don't allow the sed command to completely eliminate the filename
if [ x"$dstfile" = x ]
then
dstfile=`basename $dst`
else
true
fi
# Make a temp file name in the proper directory.
dsttmp=$dstdir/#inst.$$#
# Move or copy the file name to the temp name
$doit $instcmd $src $dsttmp &&
trap "rm -f ${dsttmp}" 0 &&
# and set any options; do chmod last to preserve setuid bits
# If any of these fail, we abort the whole thing. If we want to
# ignore errors from any of these, just make sure not to ignore
# errors from the above "$doit $instcmd $src $dsttmp" command.
if [ x"$chowncmd" != x ]; then $doit $chowncmd $dsttmp; else true;fi &&
if [ x"$chgrpcmd" != x ]; then $doit $chgrpcmd $dsttmp; else true;fi &&
if [ x"$stripcmd" != x ]; then $doit $stripcmd $dsttmp; else true;fi &&
if [ x"$chmodcmd" != x ]; then $doit $chmodcmd $dsttmp; else true;fi &&
# Now rename the file to the real destination.
$doit $rmcmd -f $dstdir/$dstfile &&
$doit $mvcmd $dsttmp $dstdir/$dstfile
fi &&
exit 0

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#
# Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
include ../Makefile.conf
TARGET=../lib/libsound.a
STARGET=../lib/libsound.so
STARGETX=../lib/libsound.so.$(SND_LIB_VERSION)
STARGETO=../lib/libsound.so.$(SND_LIB_MAJOR)
TARGETS=$(TARGET) $(STARGET)
STATIC_LIBS= control/libcontrol.a \
mixer/libmixer.a \
pcm/libpcm.a
DYNAMIC_LIBS= control/libcontrol.Sa \
mixer/libmixer.Sa \
pcm/libpcm.Sa
OBJECTS=error.o
SOBJECTS=error.So
.SUFFIXES:
.SUFFIXES: .c .s .S .o .So .a .Sa
.c.o:
$(CC) $(COPTS) $(INCLUDE) -c -o $*.o $<
.c.So:
$(CC) $(COPTS) $(INCLUDE) -fPIC -c -o $*.So $<
all: $(TARGETS)
$(TARGET): .depend $(OBJECTS) $(STATIC_LIBS)
rm -f ../lib/libsound.a
$(LINKER) -r -o $(TARGET) $(STATIC_LIBS) $(OBJECTS)
$(STARGET): .depend $(SOBJECTS) $(DYNAMIC_LIBS)
rm -f ../lib/libsound*.so*
$(CC) -shared -Wl,-soname,libsound.so.$(SND_LIB_MAJOR) $(DYNAMIC_LIBS) $(SOBJECTS) -o $(STARGETX)
ln -s libsound.so.$(SND_LIB_VERSION) $(STARGET)
ln -s libsound.so.$(SND_LIB_VERSION) $(STARGETO)
control/libcontrol.a:
$(MAKE) -C control
control/libcontrol.sa:
$(MAKE) -C control
mixer/libmixer.a:
$(MAKE) -C mixer
mixer/libmixer.sa:
$(MAKE) -C mixer
pcm/libpcm.a:
$(MAKE) -C pcm
pcm/libpcm.sa:
$(MAKE) -C pcm
clean:
$(MAKE) -C control clean
$(MAKE) -C pcm clean
$(MAKE) -C mixer clean
rm -f core .depend *.o *.So *.orig *~
rm -f ../lib/libsound.*
.depend:
$(CPP) $(COPTS) $(INCLUDE) -M *.c > .depend
#
# include a dependency file if one exists
#
ifeq (.depend,$(wildcard .depend))
include .depend
endif

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#
# Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
include ../../Makefile.conf
TARGET=libcontrol.a
STARGET=libcontrol.Sa
OBJECTS=cards.o control.o
SOBJECTS=cards.So control.So
TARGETS=$(TARGET) $(STARGET)
.SUFFIXES:
.SUFFIXES: .c .s .S .o .So .a .Sa
.c.o:
$(CC) $(COPTS) $(INCLUDE) -c -o $*.o $<
.c.So:
$(CC) $(COPTS) $(INCLUDE) -fPIC -c -o $*.So $<
all: $(TARGETS)
$(TARGET): .depend $(OBJECTS)
$(LINKER) -r -o $@ $(OBJECTS)
$(STARGET): .depend $(SOBJECTS)
$(LINKER) -r -o $@ $(SOBJECTS)
clean:
rm -f core .depend *.o *.So *.a *.Sa *.orig *~
.depend:
$(CPP) $(COPTS) $(INCLUDE) -M *.c > .depend
#
# include a dependency file if one exists
#
ifeq (.depend,$(wildcard .depend))
include .depend
endif

94
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/*
* Control Interface - main file
* Copyright (c) 1998 by Jaroslav Kysela <perex@jcu.cz>
*
*
* This library is free software; you can redistribute it and/or modify
* it under the terms of the GNU Library General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <ctype.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include "soundlib.h"
#define SND_FILE_CONTROL "/dev/sndcontrol%i"
int snd_cards( void )
{
int idx, count;
unsigned int mask;
mask = snd_cards_mask();
for ( idx = 0, count = 0; idx < SND_CARDS; idx++ ) {
if ( mask & (1 << idx) ) count++;
}
return count;
}
/*
* this routine uses very ugly method...
* need to do...
*/
unsigned int snd_cards_mask( void )
{
int fd, idx;
unsigned int mask;
char filename[32];
for ( idx = 0, mask = 0; idx < SND_CARDS; idx++ ) {
sprintf( filename, SND_FILE_CONTROL, idx );
if ( (fd = open( filename, O_RDWR )) < 0 ) continue;
close( fd );
mask |= 1 << idx;
}
return mask;
}
int snd_card_name( const char *string )
{
int card, cards;
void *handle;
struct snd_ctl_hw_info info;
cards = snd_cards();
if ( cards <= 0 ) return -ENODEV;
if ( !string ) return -EINVAL;
if ( (isdigit( *string ) && *(string+1) == 0) ||
(isdigit( *string ) && isdigit( *(string+1) ) && *(string+2) == 0) ) {
sscanf( string, "%i", &card );
card--;
if ( card < 0 || card >= cards )
return -EINVAL;
return card;
}
for ( card = 0; card < cards; card++ ) {
if ( snd_ctl_open( &handle, card ) < 0 )
continue;
if ( snd_ctl_hw_info( handle, &info ) < 0 ) {
snd_ctl_close( handle );
continue;
}
snd_ctl_close( handle );
if ( !strcmp( info.id, string ) )
return card;
}
return -ENODEV;
}

146
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/*
* Control Interface - main file
* Copyright (c) 1998 by Jaroslav Kysela <perex@jcu.cz>
*
*
* This library is free software; you can redistribute it and/or modify
* it under the terms of the GNU Library General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include "soundlib.h"
#define SND_FILE_CONTROL "/dev/sndcontrol%i"
#define SND_CTL_VERSION_MAX SND_PROTOCOL_VERSION( 1, 0, 0 )
typedef struct {
int card;
int fd;
} snd_ctl_t;
int snd_ctl_open( void **handle, int card )
{
int fd, ver;
char filename[32];
snd_ctl_t *ctl;
*handle = NULL;
if ( card < 0 || card >= SND_CARDS ) return -EINVAL;
sprintf( filename, SND_FILE_CONTROL, card );
if ( (fd = open( filename, O_RDWR )) < 0 ) return -errno;
if ( ioctl( fd, SND_CTL_IOCTL_PVERSION, &ver ) < 0 ) {
close( fd );
return -errno;
}
if ( ver > SND_CTL_VERSION_MAX ) return -SND_ERROR_UNCOMPATIBLE_VERSION;
ctl = (snd_ctl_t *)calloc( 1, sizeof( snd_ctl_t ) );
if ( ctl == NULL ) {
close( fd );
return -ENOMEM;
}
ctl -> card = card;
ctl -> fd = fd;
*handle = ctl;
return 0;
}
int snd_ctl_close( void *handle )
{
snd_ctl_t *ctl;
int res;
ctl = (snd_ctl_t *)handle;
if ( !ctl ) return -EINVAL;
res = close( ctl -> fd ) < 0 ? -errno : 0;
free( ctl );
return res;
}
int snd_ctl_file_descriptor( void *handle )
{
snd_ctl_t *ctl;
ctl = (snd_ctl_t *)handle;
if ( !ctl ) return -EINVAL;
return ctl -> fd;
}
int snd_ctl_hw_info( void *handle, struct snd_ctl_hw_info *info )
{
snd_ctl_t *ctl;
ctl = (snd_ctl_t *)handle;
if ( !ctl ) return -EINVAL;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_HW_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_ctl_pcm_info( void *handle, int dev, snd_pcm_info_t *info )
{
snd_ctl_t *ctl;
ctl = (snd_ctl_t *)handle;
if ( !ctl ) return -EINVAL;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_PCM_DEVICE, &dev ) < 0 )
return -errno;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_PCM_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_ctl_pcm_playback_info( void *handle, int dev, snd_pcm_playback_info_t *info )
{
snd_ctl_t *ctl;
ctl = (snd_ctl_t *)handle;
if ( !ctl ) return -EINVAL;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_PCM_DEVICE, &dev ) < 0 )
return -errno;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_PCM_PLAYBACK_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_ctl_pcm_record_info( void *handle, int dev, snd_pcm_record_info_t *info )
{
snd_ctl_t *ctl;
ctl = (snd_ctl_t *)handle;
if ( !ctl ) return -EINVAL;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_PCM_DEVICE, &dev ) < 0 )
return -errno;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_PCM_RECORD_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_ctl_mixer_info( void *handle, int dev, snd_mixer_info_t *info )
{
snd_ctl_t *ctl;
ctl = (snd_ctl_t *)handle;
if ( !ctl ) return -EINVAL;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_MIXER_DEVICE, &dev ) < 0 )
return -errno;
if ( ioctl( ctl -> fd, SND_CTL_IOCTL_MIXER_INFO, info ) < 0 )
return -errno;
return 0;
}

42
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/*
* Error Routines
* Copyright (c) 1998 by Jaroslav Kysela <perex@jcu.cz>
*
* snd_strerror routine needs to be recoded for locale support
*
*
* This library is free software; you can redistribute it and/or modify
* it under the terms of the GNU Library General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "soundlib.h"
static const char *snd_error_codes[] = {
"Sound protocol isn't compatible"
};
const char *snd_strerror( int errnum )
{
if ( errnum < 0 ) errnum = -errnum;
if ( errnum < SND_ERROR_BEGIN )
return (const char *)strerror( errnum );
errnum -= SND_ERROR_BEGIN;
if ( errnum >= sizeof( snd_error_codes ) / sizeof( const char * ) )
return "Unknown error";
return snd_error_codes[ errnum ];
}

41
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#
# Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
include ../../Makefile.conf
TARGET=libmixer.a
STARGET=libmixer.Sa
OBJECTS=mixer.o
SOBJECTS=mixer.So
TARGETS=$(TARGET) $(STARGET)
.SUFFIXES:
.SUFFIXES: .c .s .S .o .So .a .Sa
.c.o:
$(CC) $(COPTS) $(INCLUDE) -c -o $*.o $<
.c.So:
$(CC) $(COPTS) $(INCLUDE) -fPIC -c -o $*.So $<
all: $(TARGETS)
$(TARGET): .depend $(OBJECTS)
$(LINKER) -r -o $@ $(OBJECTS)
$(STARGET): .depend $(SOBJECTS)
$(LINKER) -r -o $@ $(SOBJECTS)
clean:
rm -f core .depend *.o *.So *.a *.Sa *.orig *~
.depend:
$(CPP) $(COPTS) $(INCLUDE) -M *.c > .depend
#
# include a dependency file if one exists
#
ifeq (.depend,$(wildcard .depend))
include .depend
endif

224
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/*
* Mixer Interface - main file
* Copyright (c) 1998 by Jaroslav Kysela <perex@jcu.cz>
*
*
* This library is free software; you can redistribute it and/or modify
* it under the terms of the GNU Library General Public License as
* published by the Free Software Foundation; either version 2 of
* the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <errno.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include "soundlib.h"
#define SND_FILE_MIXER "/dev/sndmixer%i%i"
#define SND_CTL_VERSION_MAX SND_PROTOCOL_VERSION( 1, 0, 0 )
typedef struct {
int card;
int device;
int fd;
} snd_mixer_t;
int snd_mixer_open( void **handle, int card, int device )
{
int fd, ver;
char filename[32];
snd_mixer_t *mixer;
*handle = NULL;
if ( card < 0 || card >= SND_CARDS ) return -EINVAL;
sprintf( filename, SND_FILE_MIXER, card, device );
if ( (fd = open( filename, O_RDWR )) < 0 ) return -errno;
if ( ioctl( fd, SND_MIXER_IOCTL_PVERSION, &ver ) < 0 ) {
close( fd );
return -errno;
}
if ( ver > SND_CTL_VERSION_MAX ) return -SND_ERROR_UNCOMPATIBLE_VERSION;
mixer = (snd_mixer_t *)calloc( 1, sizeof( snd_mixer_t ) );
if ( mixer == NULL ) {
close( fd );
return -ENOMEM;
}
mixer -> card = card;
mixer -> device = device;
mixer -> fd = fd;
*handle = mixer;
return 0;
}
int snd_mixer_close( void *handle )
{
snd_mixer_t *mixer;
int res;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
res = close( mixer -> fd ) < 0 ? -errno : 0;
free( mixer );
return res;
}
int snd_mixer_file_descriptor( void *handle )
{
snd_mixer_t *mixer;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
return mixer -> fd;
}
int snd_mixer_channels( void *handle )
{
snd_mixer_t *mixer;
int result;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
if ( ioctl( mixer -> fd, SND_MIXER_IOCTL_CHANNELS, &result ) < 0 )
return -errno;
return result;
}
int snd_mixer_info( void *handle, snd_mixer_info_t *info )
{
snd_mixer_t *mixer;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
if ( ioctl( mixer -> fd, SND_MIXER_IOCTL_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_mixer_exact_mode( void *handle, int enable )
{
snd_mixer_t *mixer;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
if ( ioctl( mixer -> fd, SND_MIXER_IOCTL_EXACT, &enable ) < 0 )
return -errno;
return 0;
}
int snd_mixer_channel( void *handle, const char *channel_id )
{
snd_mixer_t *mixer;
snd_mixer_channel_info_t info;
int idx, channels, err;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
/* bellow implementation isn't optimized for speed */
/* info about channels should be cached in the snd_mixer_t structure */
if ( (channels = snd_mixer_channels( handle )) < 0 )
return channels;
for ( idx = 0; idx < channels; idx++ ) {
if ( (err = snd_mixer_channel_info( handle, idx, &info )) < 0 )
return err;
if ( !strncmp( channel_id, info.name, sizeof( info.name ) ) )
return idx;
}
return -EINVAL;
}
int snd_mixer_channel_info( void *handle, int channel, snd_mixer_channel_info_t *info )
{
snd_mixer_t *mixer;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
info -> channel = channel;
if ( ioctl( mixer -> fd, SND_MIXER_IOCTL_CHANNEL_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_mixer_channel_read( void *handle, int channel, snd_mixer_channel_t *data )
{
snd_mixer_t *mixer;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
data -> channel = channel;
if ( ioctl( mixer -> fd, SND_MIXER_IOCTL_CHANNEL_READ, data ) < 0 )
return -errno;
return 0;
}
int snd_mixer_channel_write( void *handle, int channel, snd_mixer_channel_t *data )
{
snd_mixer_t *mixer;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
data -> channel = channel;
if ( ioctl( mixer -> fd, SND_MIXER_IOCTL_CHANNEL_WRITE, data ) < 0 )
return -errno;
return 0;
}
int snd_mixer_special_read( void *handle, snd_mixer_special_t *special )
{
snd_mixer_t *mixer;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
if ( ioctl( mixer -> fd, SND_MIXER_IOCTL_SPECIAL_READ, special ) < 0 )
return -errno;
return 0;
}
int snd_mixer_special_write( void *handle, snd_mixer_special_t *special )
{
snd_mixer_t *mixer;
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
if ( ioctl( mixer -> fd, SND_MIXER_IOCTL_SPECIAL_WRITE, special ) < 0 )
return -errno;
return 0;
}
int snd_mixer_read( void *handle, snd_mixer_callbacks_t *callbacks )
{
snd_mixer_t *mixer;
int idx, result, count;
unsigned int cmd, tmp;
unsigned char buffer[ 64 ];
mixer = (snd_mixer_t *)handle;
if ( !mixer ) return -EINVAL;
count = 0;
while ( (result = read( mixer -> fd, &buffer, sizeof( buffer ))) > 0 ) {
if ( result & 7 ) return -EIO;
if ( !callbacks ) continue;
for ( idx = 0; idx < result; idx += 8 ) {
cmd = *(unsigned int *)&buffer[ idx ];
tmp = *(unsigned int *)&buffer[ idx + 4 ];
if ( cmd == 0 && callbacks -> channel_was_changed ) {
callbacks -> channel_was_changed( callbacks -> private_data, (int)tmp );
}
}
count += result >> 3; /* return only number of changes */
}
return result >= 0 ? count : -errno;
}

40
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#
# Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
include ../../Makefile.conf
TARGET=libpcm.a
STARGET=libpcm.Sa
OBJECTS=pcm.o
SOBJECTS=pcm.So
TARGETS=$(TARGET) $(STARGET)
.SUFFIXES: .c .s .S .o .So .a .Sa
.c.o:
$(CC) $(COPTS) $(INCLUDE) -c -o $*.o $<
.c.So:
$(CC) $(COPTS) $(INCLUDE) -fPIC -c -o $*.So $<
all: $(TARGETS)
$(TARGET): .depend $(OBJECTS)
$(LINKER) -r -o $@ $(OBJECTS)
$(STARGET): .depend $(SOBJECTS)
$(LINKER) -r -o $@ $(SOBJECTS)
clean:
rm -f core .depend *.o *.So *.a *.Sa *.orig *~
.depend:
$(CPP) $(COPTS) $(INCLUDE) -M *.c > .depend
#
# include a dependency file if one exists
#
ifeq (.depend,$(wildcard .depend))
include .depend
endif

265
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/*
* PCM Interface - main file
* Copyright (c) 1998 by Jaroslav Kysela <perex@jcu.cz>
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include "soundlib.h"
#define SND_FILE_PCM "/dev/sndpcm%i%i"
#define SND_CTL_VERSION_MAX SND_PROTOCOL_VERSION( 1, 0, 0 )
typedef struct {
int card;
int device;
int fd;
} snd_pcm_t;
int snd_pcm_open( void **handle, int card, int device, int mode )
{
int fd, ver;
char filename[32];
snd_pcm_t *pcm;
*handle = NULL;
if ( card < 0 || card >= SND_CARDS ) return -EINVAL;
sprintf( filename, SND_FILE_PCM, card, device );
if ( (fd = open( filename, mode )) < 0 ) return -errno;
if ( ioctl( fd, SND_PCM_IOCTL_PVERSION, &ver ) < 0 ) {
close( fd );
return -errno;
}
if ( ver > SND_CTL_VERSION_MAX ) return -SND_ERROR_UNCOMPATIBLE_VERSION;
pcm = (snd_pcm_t *)calloc( 1, sizeof( snd_pcm_t ) );
if ( pcm == NULL ) {
close( fd );
return -ENOMEM;
}
pcm -> card = card;
pcm -> device = device;
pcm -> fd = fd;
*handle = pcm;
return 0;
}
int snd_pcm_close( void *handle )
{
snd_pcm_t *pcm;
int res;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
res = close( pcm -> fd ) < 0 ? -errno : 0;
free( pcm );
return res;
}
int snd_pcm_file_descriptor( void *handle )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
return pcm -> fd;
}
int snd_pcm_block_mode( void *handle, int enable )
{
snd_pcm_t *pcm;
long flags;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( fcntl( pcm -> fd, F_GETFL, &flags ) < 0 )
return -errno;
if ( enable )
flags |= O_NONBLOCK;
else
flags &= ~O_NONBLOCK;
if ( fcntl( pcm -> fd, F_SETFL, &flags ) < 0 )
return -errno;
return 0;
}
int snd_pcm_info( void *handle, snd_pcm_info_t *info )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_pcm_playback_info( void *handle, snd_pcm_playback_info_t *info )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_PLAYBACK_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_pcm_record_info( void *handle, snd_pcm_record_info_t *info )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_RECORD_INFO, info ) < 0 )
return -errno;
return 0;
}
int snd_pcm_playback_format( void *handle, snd_pcm_format_t *format )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_PLAYBACK_FORMAT, format ) < 0 )
return -errno;
return 0;
}
int snd_pcm_record_format( void *handle, snd_pcm_format_t *format )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_RECORD_FORMAT, format ) < 0 )
return -errno;
return 0;
}
int snd_pcm_playback_params( void *handle, snd_pcm_playback_params_t *params )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_PLAYBACK_PARAMS, params ) < 0 )
return -errno;
return 0;
}
int snd_pcm_record_params( void *handle, snd_pcm_record_params_t *params )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_RECORD_PARAMS, params ) < 0 )
return -errno;
return 0;
}
int snd_pcm_playback_status( void *handle, snd_pcm_playback_status_t *status )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_PLAYBACK_STATUS, status ) < 0 )
return -errno;
return 0;
}
int snd_pcm_record_status( void *handle, snd_pcm_record_status_t *status )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_RECORD_STATUS, status ) < 0 )
return -errno;
return 0;
}
int snd_pcm_drain_playback( void *handle )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_DRAIN_PLAYBACK ) < 0 )
return -errno;
return 0;
}
int snd_pcm_flush_playback( void *handle )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_FLUSH_PLAYBACK ) < 0 )
return -errno;
return 0;
}
int snd_pcm_flush_record( void *handle )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_FLUSH_RECORD ) < 0 )
return -errno;
return 0;
}
int snd_pcm_playback_time( void *handle, int enable )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_PLAYBACK_TIME, &enable ) < 0 )
return -errno;
return 0;
}
int snd_pcm_record_time( void *handle, int enable )
{
snd_pcm_t *pcm;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
if ( ioctl( pcm -> fd, SND_PCM_IOCTL_RECORD_TIME, &enable ) < 0 )
return -errno;
return 0;
}
ssize_t snd_pcm_write( void *handle, const void *buffer, size_t size )
{
snd_pcm_t *pcm;
ssize_t result;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
result = write( pcm -> fd, buffer, size );
if ( result < 0 ) return -errno;
return result;
}
ssize_t snd_pcm_read( void *handle, void *buffer, size_t size )
{
snd_pcm_t *pcm;
ssize_t result;
pcm = (snd_pcm_t *)handle;
if ( !pcm ) return -EINVAL;
result = read( pcm -> fd, buffer, size );
if ( result < 0 ) return -errno;
return result;
}

15
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@ -0,0 +1,15 @@
CC = gcc
CFLAGS = -static -O2 -g -Wall -pipe
TARGETS = control mixer
LIB = -L../lib -lsound
all: $(TARGETS)
control: control.c
$(CC) $(CFLAGS) $(LIB) -o control control.c
mixer: mixer.c
$(CC) $(CFLAGS) $(LIB) -o mixer mixer.c
clean:
rm -f *.o $(TARGETS) *~

67
test/control.c Normal file
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#include <stdio.h>
#include <string.h>
#include "../include/soundlib.h"
void main( void )
{
int idx, idx1, cards, err;
void *handle;
struct snd_ctl_hw_info info;
snd_pcm_info_t pcminfo;
snd_mixer_info_t mixerinfo;
char str[128];
cards = snd_cards();
printf( "Detected %i soundcard%s...\n", cards, cards > 1 ? "s" : "" );
if ( cards <= 0 ) {
printf( "Giving up...\n" );
return;
}
for ( idx = 0; idx < cards; idx++ ) {
if ( (err = snd_ctl_open( &handle, idx )) < 0 ) {
printf( "Open error: %s\n", snd_strerror( err ) );
continue;
}
if ( (err = snd_ctl_hw_info( handle, &info )) < 0 ) {
printf( "HW info error: %s\n", snd_strerror( err ) );
continue;
}
printf( "Soundcard #%i:\n", idx + 1 );
printf( " type - %i\n", info.type );
printf( " gcaps - 0x%x\n", info.gcaps );
printf( " lcaps - 0x%x\n", info.lcaps );
printf( " pcm devs - 0x%x\n", info.pcmdevs );
printf( " mixer devs - 0x%x\n", info.mixerdevs );
printf( " midi devs - 0x%x\n", info.mididevs );
memset( str, 0, sizeof( str ) );
strncpy( str, info.id, sizeof( info.id ) );
printf( " id - '%s'\n", str );
printf( " abbreviation - '%s'\n", info.abbreviation );
printf( " name - '%s'\n", info.name );
printf( " longname - '%s'\n", info.longname );
for ( idx1 = 0; idx1 < info.pcmdevs; idx1++ ) {
printf( "PCM info, device #%i:\n", idx1 );
if ( (err = snd_ctl_pcm_info( handle, idx1, &pcminfo )) < 0 ) {
printf( " PCM info error: %s\n", snd_strerror( err ) );
continue;
}
printf( " type - %i\n", pcminfo.type );
printf( " flags - 0x%x\n", pcminfo.flags );
printf( " id - '%s'\n", pcminfo.id );
printf( " name - '%s'\n", pcminfo.name );
}
for ( idx1 = 0; idx1 < info.mixerdevs; idx1++ ) {
printf( "MIXER info, device #%i:\n", idx1 );
if ( (err = snd_ctl_mixer_info( handle, idx1, &mixerinfo )) < 0 ) {
printf( " MIXER info error: %s\n", snd_strerror( err ) );
continue;
}
printf( " type - %i\n", mixerinfo.type );
printf( " channels - %i\n", mixerinfo.channels );
printf( " caps - 0x%x\n", mixerinfo.caps );
printf( " id - '%s'\n", mixerinfo.id );
printf( " name - '%s'\n", mixerinfo.name );
}
snd_ctl_close( handle );
}
}

60
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#include <stdio.h>
#include <string.h>
#include "../include/soundlib.h"
static void mixer_test( int card, int device )
{
int err, channels;
void *handle;
snd_mixer_info_t info;
if ( (err = snd_mixer_open( &handle, card, device)) < 0 ) {
printf( "Mixer open error: %s\n", snd_strerror( err ) );
return;
}
printf( "Mixer %i/%i open ok...\n", card, device );
channels = snd_mixer_channels( handle );
if ( channels < 0 ) {
printf( "Mixer channels error: %s\n", snd_strerror( channels ) );
return;
}
printf( " Mixer have %i channel%s.\n", channels, channels > 1 ? "s" : "" );
if ( (err = snd_mixer_info( handle, &info )) < 0 ) {
printf( "Mixer info error: %s\n", snd_strerror( err ) );
return;
}
printf( " Info:\n" );
printf( " type - %i\n", info.type );
printf( " channels - %i\n", info.channels );
printf( " caps - 0x%x\n", info.caps );
printf( " id - '%s'\n", info.id );
printf( " name - '%s'\n", info.name );
snd_mixer_close( handle );
}
void main( void )
{
int idx, idx1, cards, err;
void *handle;
struct snd_ctl_hw_info info;
cards = snd_cards();
printf( "Detected %i soundcard%s...\n", cards, cards > 1 ? "s" : "" );
if ( cards <= 0 ) {
printf( "Giving up...\n" );
return;
}
for ( idx = 0; idx < cards; idx++ ) {
if ( (err = snd_ctl_open( &handle, idx )) < 0 ) {
printf( "Open error: %s\n", snd_strerror( err ) );
continue;
}
if ( (err = snd_ctl_hw_info( handle, &info )) < 0 ) {
printf( "HW info error: %s\n", snd_strerror( err ) );
continue;
}
for ( idx1 = 0; idx1 < info.mixerdevs; idx1++ )
mixer_test( idx, idx1 );
snd_ctl_close( handle );
}
}

10
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#
# Makefile for ALSA library
# Copyright (c) 1994-98 by Jaroslav Kysela <perex@jcu.cz>
#
include ../Makefile.conf
clean:
rm -f core .depend *.o *.orig *~

77
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%define ver 0.0.9
%define rel 1
%define prefix /
Summary: Advanced Linux Sound Architecture (ALSA) - Library
Name: alsa-lib
Version: %ver
Release: %rel
Copyright: GPL
Group: System/Libraries
Source: ftp://alsa.jcu.cz/pub/lib/alsa-lib-%{ver}.tar.gz
BuildRoot: /tmp/alsa-lib-%{ver}
Packager: Helge Jensen <slog@slog.dk>
URL: http://alsa.jcu.cz
Docdir: %{prefix}/usr/doc
%description
Advanced Linux Sound Architecture (ALSA) - Library
Features
========
* general
- modularized architecture with support for 2.0 and latest 2.1 kernels
- support for versioned and exported symbols
- full proc filesystem support - /proc/sound
* ISA soundcards
- support for 128k ISA DMA buffer
* mixer
- new enhanced API for applications
- support for unlimited number of channels
- volume can be set in three ways (percentual (0-100), exact and decibel)
- support for mute (and hardware mute if hardware supports it)
- support for mixer events
- this allows two or more applications to be synchronized
* digital audio (PCM)
- new enhanced API for applications
- full real duplex support
- full duplex support for SoundBlaster 16/AWE soundcards
- digital audio data for playback and record should be read back using
proc filesystem
* OSS/Lite compatibility
- full mixer compatibity
- full PCM (/dev/dsp) compatibility
%changelog
* Mon May 28 1998 Helge Jensen <slog@slog.dk>
- Made SPEC file
%prep
%setup -n alsa-lib
%build
./configure --prefix=/usr
make
%install
mkdir -p $RPM_BUILD_ROOT/usr/include/sys
mkdir -p $RPM_BUILD_ROOT/usr/lib
make prefix="$RPM_BUILD_ROOT/usr" install
%clean
rm -rf $RPM_BUILD_ROOT
%files
%defattr(-, root, root)
%doc doc/*.html
%doc doc/*.sgml
%doc doc/*.txt
%{prefix}/usr/include/sys/soundlib.h
%{prefix}/usr/lib/lib*

78
utils/alsa-lib.spec.in Normal file
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%define ver @SND_LIB_VERSION@
%define rel 1
%define prefix /
Summary: Advanced Linux Sound Architecture (ALSA) - Library
Name: alsa-lib
Version: %ver
Release: %rel
Copyright: GPL
Group: System/Libraries
Source: ftp://alsa.jcu.cz/pub/lib/alsa-lib-%{ver}.tar.gz
BuildRoot: /tmp/alsa-lib-%{ver}
Packager: Helge Jensen <slog@slog.dk>
URL: http://alsa.jcu.cz
Docdir: %{prefix}/usr/doc
Requires: alsa-driver
%description
Advanced Linux Sound Architecture (ALSA) - Library
Features
========
* general
- modularized architecture with support for 2.0 and latest 2.1 kernels
- support for versioned and exported symbols
- full proc filesystem support - /proc/sound
* ISA soundcards
- support for 128k ISA DMA buffer
* mixer
- new enhanced API for applications
- support for unlimited number of channels
- volume can be set in three ways (percentual (0-100), exact and decibel)
- support for mute (and hardware mute if hardware supports it)
- support for mixer events
- this allows two or more applications to be synchronized
* digital audio (PCM)
- new enhanced API for applications
- full real duplex support
- full duplex support for SoundBlaster 16/AWE soundcards
- digital audio data for playback and record should be read back using
proc filesystem
* OSS/Lite compatibility
- full mixer compatibity
- full PCM (/dev/dsp) compatibility
%changelog
* Mon May 28 1998 Helge Jensen <slog@slog.dk>
- Made SPEC file
%prep
%setup -n alsa-lib
%build
./configure --prefix=/usr
make
%install
mkdir -p $RPM_BUILD_ROOT/usr/include/sys
mkdir -p $RPM_BUILD_ROOT/usr/lib
make prefix="$RPM_BUILD_ROOT/usr" install
%clean
rm -rf $RPM_BUILD_ROOT
%files
%defattr(-, root, root)
%doc doc/*.html
%doc doc/*.sgml
%doc doc/*.txt
%{prefix}/usr/include/sys/soundlib.h
%{prefix}/usr/lib/lib*

35
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#!/bin/bash
source=.
version=`cat $source/../version`
package=$source/../../alsa-lib-$version.tar.gz
if [ ! -r $package ]; then
echo "Error: wrong package: $package"
exit 1
fi
make -C .. pack
cp -fv $package /usr/src/redhat/SOURCES
if [ ! -r $source/buildrpm ]; then
echo "Error: invalid directory: $source"
exit 1
fi
if [ ! -d /usr/src/redhat ]; then
echo "Error: /usr/src/redhat directory not found"
exit 1
fi
if [ ! -r $source/alsa-lib.spec ]; then
cd $source/..
./configure
cd utils
fi
cp -fv $source/alsa-lib.spec /usr/src/redhat/SPECS
cd /usr/src/redhat/SPECS
rpm -ba alsa-lib.spec
cd /usr/src/redhat

1
version Normal file
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@ -0,0 +1 @@
0.0.9