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" >&6; } if ${lt_cv_prog_gnu_ld+:} false; then : $as_echo_n "(cached) " >&6 else # I'd rather use --version here, but apparently some GNU lds only accept -v. case `$LD -v 2>&1 &5 $as_echo "$lt_cv_prog_gnu_ld" >&6; } with_gnu_ld=$lt_cv_prog_gnu_ld { $as_echo "$as_me:${as_lineno-$LINENO}: checking for BSD- or MS-compatible name lister (nm)" >&5 $as_echo_n "checking for BSD- or MS-compatible name lister (nm)... " >&6; } if ${lt_cv_path_NM+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$NM"; then # Let the user override the test. lt_cv_path_NM=$NM else lt_nm_to_check=${ac_tool_prefix}nm if test -n "$ac_tool_prefix" && test "$build" = "$host"; then lt_nm_to_check="$lt_nm_to_check nm" fi for lt_tmp_nm in $lt_nm_to_check; do lt_save_ifs=$IFS; IFS=$PATH_SEPARATOR for ac_dir in $PATH /usr/ccs/bin/elf /usr/ccs/bin /usr/ucb /bin; do IFS=$lt_save_ifs test -z "$ac_dir" && ac_dir=. tmp_nm=$ac_dir/$lt_tmp_nm if test -f "$tmp_nm" || test -f "$tmp_nm$ac_exeext"; then # Check to see if the nm accepts a BSD-compat flag. # Adding the 'sed 1q' prevents false positives on HP-UX, which says: # nm: unknown option "B" ignored # Tru64's nm complains that /dev/null is an invalid object file # MSYS converts /dev/null to NUL, MinGW nm treats NUL as empty case $build_os in mingw*) lt_bad_file=conftest.nm/nofile ;; *) lt_bad_file=/dev/null ;; esac case `"$tmp_nm" -B $lt_bad_file 2>&1 | sed '1q'` in *$lt_bad_file* | *'Invalid file or object type'*) lt_cv_path_NM="$tmp_nm -B" break 2 ;; *) case `"$tmp_nm" -p /dev/null 2>&1 | sed '1q'` in */dev/null*) lt_cv_path_NM="$tmp_nm -p" break 2 ;; *) lt_cv_path_NM=${lt_cv_path_NM="$tmp_nm"} # keep the first match, but continue # so that we can try to find one that supports BSD flags ;; esac ;; esac fi done IFS=$lt_save_ifs done : ${lt_cv_path_NM=no} fi fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_path_NM" >&5 $as_echo "$lt_cv_path_NM" >&6; } if test no != "$lt_cv_path_NM"; then NM=$lt_cv_path_NM else # Didn't find any BSD compatible name lister, look for dumpbin. if test -n "$DUMPBIN"; then : # Let the user override the test. else if test -n "$ac_tool_prefix"; then for ac_prog in dumpbin "link -dump" do # Extract the first word of "$ac_tool_prefix$ac_prog", so it can be a program name with args. set dummy $ac_tool_prefix$ac_prog; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_DUMPBIN+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$DUMPBIN"; then ac_cv_prog_DUMPBIN="$DUMPBIN" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_DUMPBIN="$ac_tool_prefix$ac_prog" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi DUMPBIN=$ac_cv_prog_DUMPBIN if test -n "$DUMPBIN"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $DUMPBIN" >&5 $as_echo "$DUMPBIN" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi test -n "$DUMPBIN" && break done fi if test -z "$DUMPBIN"; then ac_ct_DUMPBIN=$DUMPBIN for ac_prog in dumpbin "link -dump" do # Extract the first word of "$ac_prog", so it can be a program name with args. set dummy $ac_prog; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_DUMPBIN+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_DUMPBIN"; then ac_cv_prog_ac_ct_DUMPBIN="$ac_ct_DUMPBIN" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_DUMPBIN="$ac_prog" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_DUMPBIN=$ac_cv_prog_ac_ct_DUMPBIN if test -n "$ac_ct_DUMPBIN"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_DUMPBIN" >&5 $as_echo "$ac_ct_DUMPBIN" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi test -n "$ac_ct_DUMPBIN" && break done if test "x$ac_ct_DUMPBIN" = x; then DUMPBIN=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac DUMPBIN=$ac_ct_DUMPBIN fi fi case `$DUMPBIN -symbols -headers /dev/null 2>&1 | sed '1q'` in *COFF*) DUMPBIN="$DUMPBIN -symbols -headers" ;; *) DUMPBIN=: ;; esac fi if test : != "$DUMPBIN"; then NM=$DUMPBIN fi fi test -z "$NM" && NM=nm { $as_echo "$as_me:${as_lineno-$LINENO}: checking the name lister ($NM) interface" >&5 $as_echo_n "checking the name lister ($NM) interface... " >&6; } if ${lt_cv_nm_interface+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_nm_interface="BSD nm" echo "int some_variable = 0;" > conftest.$ac_ext (eval echo "\"\$as_me:$LINENO: $ac_compile\"" >&5) (eval "$ac_compile" 2>conftest.err) cat conftest.err >&5 (eval echo "\"\$as_me:$LINENO: $NM \\\"conftest.$ac_objext\\\"\"" >&5) (eval "$NM \"conftest.$ac_objext\"" 2>conftest.err > conftest.out) cat conftest.err >&5 (eval echo "\"\$as_me:$LINENO: output\"" >&5) cat conftest.out >&5 if $GREP 'External.*some_variable' conftest.out > /dev/null; then lt_cv_nm_interface="MS dumpbin" fi rm -f conftest* fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_nm_interface" >&5 $as_echo "$lt_cv_nm_interface" >&6; } { $as_echo "$as_me:${as_lineno-$LINENO}: checking whether ln -s works" >&5 $as_echo_n "checking whether ln -s works... 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Even if it were fixed, the result of this # check would be larger than it should be. lt_cv_sys_max_cmd_len=12288; # 12K is about right ;; gnu*) # Under GNU Hurd, this test is not required because there is # no limit to the length of command line arguments. # Libtool will interpret -1 as no limit whatsoever lt_cv_sys_max_cmd_len=-1; ;; cygwin* | mingw* | cegcc*) # On Win9x/ME, this test blows up -- it succeeds, but takes # about 5 minutes as the teststring grows exponentially. # Worse, since 9x/ME are not pre-emptively multitasking, # you end up with a "frozen" computer, even though with patience # the test eventually succeeds (with a max line length of 256k). # Instead, let's just punt: use the minimum linelength reported by # all of the supported platforms: 8192 (on NT/2K/XP). lt_cv_sys_max_cmd_len=8192; ;; mint*) # On MiNT this can take a long time and run out of memory. lt_cv_sys_max_cmd_len=8192; ;; amigaos*) # On AmigaOS with pdksh, this test takes hours, literally. # So we just punt and use a minimum line length of 8192. lt_cv_sys_max_cmd_len=8192; ;; bitrig* | darwin* | dragonfly* | freebsd* | netbsd* | openbsd*) # This has been around since 386BSD, at least. Likely further. if test -x /sbin/sysctl; then lt_cv_sys_max_cmd_len=`/sbin/sysctl -n kern.argmax` elif test -x /usr/sbin/sysctl; then lt_cv_sys_max_cmd_len=`/usr/sbin/sysctl -n kern.argmax` else lt_cv_sys_max_cmd_len=65536 # usable default for all BSDs fi # And add a safety zone lt_cv_sys_max_cmd_len=`expr $lt_cv_sys_max_cmd_len \/ 4` lt_cv_sys_max_cmd_len=`expr $lt_cv_sys_max_cmd_len \* 3` ;; interix*) # We know the value 262144 and hardcode it with a safety zone (like BSD) lt_cv_sys_max_cmd_len=196608 ;; os2*) # The test takes a long time on OS/2. lt_cv_sys_max_cmd_len=8192 ;; osf*) # Dr. Hans Ekkehard Plesser reports seeing a kernel panic running configure # due to this test when exec_disable_arg_limit is 1 on Tru64. It is not # nice to cause kernel panics so lets avoid the loop below. # First set a reasonable default. lt_cv_sys_max_cmd_len=16384 # if test -x /sbin/sysconfig; then case `/sbin/sysconfig -q proc exec_disable_arg_limit` in *1*) lt_cv_sys_max_cmd_len=-1 ;; esac fi ;; sco3.2v5*) lt_cv_sys_max_cmd_len=102400 ;; sysv5* | sco5v6* | sysv4.2uw2*) kargmax=`grep ARG_MAX /etc/conf/cf.d/stune 2>/dev/null` if test -n "$kargmax"; then lt_cv_sys_max_cmd_len=`echo $kargmax | sed 's/.*[ ]//'` else lt_cv_sys_max_cmd_len=32768 fi ;; *) lt_cv_sys_max_cmd_len=`(getconf ARG_MAX) 2> /dev/null` if test -n "$lt_cv_sys_max_cmd_len" && \ test undefined != "$lt_cv_sys_max_cmd_len"; then lt_cv_sys_max_cmd_len=`expr $lt_cv_sys_max_cmd_len \/ 4` lt_cv_sys_max_cmd_len=`expr $lt_cv_sys_max_cmd_len \* 3` else # Make teststring a little bigger before we do anything with it. # a 1K string should be a reasonable start. for i in 1 2 3 4 5 6 7 8; do teststring=$teststring$teststring done SHELL=${SHELL-${CONFIG_SHELL-/bin/sh}} # If test is not a shell built-in, we'll probably end up computing a # maximum length that is only half of the actual maximum length, but # we can't tell. while { test X`env echo "$teststring$teststring" 2>/dev/null` \ = "X$teststring$teststring"; } >/dev/null 2>&1 && test 17 != "$i" # 1/2 MB should be enough do i=`expr $i + 1` teststring=$teststring$teststring done # Only check the string length outside the loop. lt_cv_sys_max_cmd_len=`expr "X$teststring" : ".*" 2>&1` teststring= # Add a significant safety factor because C++ compilers can tack on # massive amounts of additional arguments before passing them to the # linker. It appears as though 1/2 is a usable value. lt_cv_sys_max_cmd_len=`expr $lt_cv_sys_max_cmd_len \/ 2` fi ;; esac fi if test -n "$lt_cv_sys_max_cmd_len"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_sys_max_cmd_len" >&5 $as_echo "$lt_cv_sys_max_cmd_len" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: none" >&5 $as_echo "none" >&6; } fi max_cmd_len=$lt_cv_sys_max_cmd_len : ${CP="cp -f"} : ${MV="mv -f"} : ${RM="rm -f"} if ( (MAIL=60; unset MAIL) || exit) >/dev/null 2>&1; then lt_unset=unset else lt_unset=false fi # test EBCDIC or ASCII case `echo X|tr X '\101'` in A) # ASCII based system # \n is not interpreted correctly by Solaris 8 /usr/ucb/tr lt_SP2NL='tr \040 \012' lt_NL2SP='tr \015\012 \040\040' ;; *) # EBCDIC based system lt_SP2NL='tr \100 \n' lt_NL2SP='tr \r\n \100\100' ;; esac { $as_echo "$as_me:${as_lineno-$LINENO}: checking how to convert $build file names to $host format" >&5 $as_echo_n "checking how to convert $build file names to $host format... " >&6; } if ${lt_cv_to_host_file_cmd+:} false; then : $as_echo_n "(cached) " >&6 else case $host in *-*-mingw* ) case $build in *-*-mingw* ) # actually msys lt_cv_to_host_file_cmd=func_convert_file_msys_to_w32 ;; *-*-cygwin* ) lt_cv_to_host_file_cmd=func_convert_file_cygwin_to_w32 ;; * ) # otherwise, assume *nix lt_cv_to_host_file_cmd=func_convert_file_nix_to_w32 ;; esac ;; *-*-cygwin* ) case $build in *-*-mingw* ) # actually msys lt_cv_to_host_file_cmd=func_convert_file_msys_to_cygwin ;; *-*-cygwin* ) lt_cv_to_host_file_cmd=func_convert_file_noop ;; * ) # otherwise, assume *nix lt_cv_to_host_file_cmd=func_convert_file_nix_to_cygwin ;; esac ;; * ) # unhandled hosts (and "normal" native builds) lt_cv_to_host_file_cmd=func_convert_file_noop ;; esac fi to_host_file_cmd=$lt_cv_to_host_file_cmd { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_to_host_file_cmd" >&5 $as_echo "$lt_cv_to_host_file_cmd" >&6; } { $as_echo "$as_me:${as_lineno-$LINENO}: checking how to convert $build file names to toolchain format" >&5 $as_echo_n "checking how to convert $build file names to toolchain format... 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" >&6; } if ${lt_cv_ld_reload_flag+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_ld_reload_flag='-r' fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_ld_reload_flag" >&5 $as_echo "$lt_cv_ld_reload_flag" >&6; } reload_flag=$lt_cv_ld_reload_flag case $reload_flag in "" | " "*) ;; *) reload_flag=" $reload_flag" ;; esac reload_cmds='$LD$reload_flag -o $output$reload_objs' case $host_os in cygwin* | mingw* | pw32* | cegcc*) if test yes != "$GCC"; then reload_cmds=false fi ;; darwin*) if test yes = "$GCC"; then reload_cmds='$LTCC $LTCFLAGS -nostdlib $wl-r -o $output$reload_objs' else reload_cmds='$LD$reload_flag -o $output$reload_objs' fi ;; esac if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}objdump", so it can be a program name with args. set dummy ${ac_tool_prefix}objdump; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_OBJDUMP+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$OBJDUMP"; then ac_cv_prog_OBJDUMP="$OBJDUMP" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_OBJDUMP="${ac_tool_prefix}objdump" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi OBJDUMP=$ac_cv_prog_OBJDUMP if test -n "$OBJDUMP"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $OBJDUMP" >&5 $as_echo "$OBJDUMP" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_OBJDUMP"; then ac_ct_OBJDUMP=$OBJDUMP # Extract the first word of "objdump", so it can be a program name with args. set dummy objdump; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_OBJDUMP+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_OBJDUMP"; then ac_cv_prog_ac_ct_OBJDUMP="$ac_ct_OBJDUMP" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_OBJDUMP="objdump" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_OBJDUMP=$ac_cv_prog_ac_ct_OBJDUMP if test -n "$ac_ct_OBJDUMP"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_OBJDUMP" >&5 $as_echo "$ac_ct_OBJDUMP" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_OBJDUMP" = x; then OBJDUMP="false" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac OBJDUMP=$ac_ct_OBJDUMP fi else OBJDUMP="$ac_cv_prog_OBJDUMP" fi test -z "$OBJDUMP" && OBJDUMP=objdump { $as_echo "$as_me:${as_lineno-$LINENO}: checking how to recognize dependent libraries" >&5 $as_echo_n "checking how to recognize dependent libraries... " >&6; } if ${lt_cv_deplibs_check_method+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_file_magic_cmd='$MAGIC_CMD' lt_cv_file_magic_test_file= lt_cv_deplibs_check_method='unknown' # Need to set the preceding variable on all platforms that support # interlibrary dependencies. # 'none' -- dependencies not supported. # 'unknown' -- same as none, but documents that we really don't know. # 'pass_all' -- all dependencies passed with no checks. # 'test_compile' -- check by making test program. # 'file_magic [[regex]]' -- check by looking for files in library path # that responds to the $file_magic_cmd with a given extended regex. # If you have 'file' or equivalent on your system and you're not sure # whether 'pass_all' will *always* work, you probably want this one. case $host_os in aix[4-9]*) lt_cv_deplibs_check_method=pass_all ;; beos*) lt_cv_deplibs_check_method=pass_all ;; bsdi[45]*) lt_cv_deplibs_check_method='file_magic ELF [0-9][0-9]*-bit [ML]SB (shared object|dynamic lib)' lt_cv_file_magic_cmd='/usr/bin/file -L' lt_cv_file_magic_test_file=/shlib/libc.so ;; cygwin*) # func_win32_libid is a shell function defined in ltmain.sh lt_cv_deplibs_check_method='file_magic ^x86 archive import|^x86 DLL' lt_cv_file_magic_cmd='func_win32_libid' ;; mingw* | pw32*) # Base MSYS/MinGW do not provide the 'file' command needed by # func_win32_libid shell function, so use a weaker test based on 'objdump', # unless we find 'file', for example because we are cross-compiling. if ( file / ) >/dev/null 2>&1; then lt_cv_deplibs_check_method='file_magic ^x86 archive import|^x86 DLL' lt_cv_file_magic_cmd='func_win32_libid' else # Keep this pattern in sync with the one in func_win32_libid. lt_cv_deplibs_check_method='file_magic file format (pei*-i386(.*architecture: i386)?|pe-arm-wince|pe-x86-64)' lt_cv_file_magic_cmd='$OBJDUMP -f' fi ;; cegcc*) # use the weaker test based on 'objdump'. See mingw*. lt_cv_deplibs_check_method='file_magic file format pe-arm-.*little(.*architecture: arm)?' lt_cv_file_magic_cmd='$OBJDUMP -f' ;; darwin* | rhapsody*) lt_cv_deplibs_check_method=pass_all ;; freebsd* | dragonfly*) if echo __ELF__ | $CC -E - | $GREP __ELF__ > /dev/null; then case $host_cpu in i*86 ) # Not sure whether the presence of OpenBSD here was a mistake. # Let's accept both of them until this is cleared up. lt_cv_deplibs_check_method='file_magic (FreeBSD|OpenBSD|DragonFly)/i[3-9]86 (compact )?demand paged shared library' lt_cv_file_magic_cmd=/usr/bin/file lt_cv_file_magic_test_file=`echo /usr/lib/libc.so.*` ;; esac else lt_cv_deplibs_check_method=pass_all fi ;; haiku*) lt_cv_deplibs_check_method=pass_all ;; hpux10.20* | hpux11*) lt_cv_file_magic_cmd=/usr/bin/file case $host_cpu in ia64*) lt_cv_deplibs_check_method='file_magic (s[0-9][0-9][0-9]|ELF-[0-9][0-9]) shared object file - IA64' lt_cv_file_magic_test_file=/usr/lib/hpux32/libc.so ;; hppa*64*) lt_cv_deplibs_check_method='file_magic (s[0-9][0-9][0-9]|ELF[ -][0-9][0-9])(-bit)?( [LM]SB)? shared object( file)?[, -]* PA-RISC [0-9]\.[0-9]' lt_cv_file_magic_test_file=/usr/lib/pa20_64/libc.sl ;; *) lt_cv_deplibs_check_method='file_magic (s[0-9][0-9][0-9]|PA-RISC[0-9]\.[0-9]) shared library' lt_cv_file_magic_test_file=/usr/lib/libc.sl ;; esac ;; interix[3-9]*) # PIC code is broken on Interix 3.x, that's why |\.a not |_pic\.a here lt_cv_deplibs_check_method='match_pattern /lib[^/]+(\.so|\.a)$' ;; irix5* | irix6* | nonstopux*) case $LD in *-32|*"-32 ") libmagic=32-bit;; *-n32|*"-n32 ") libmagic=N32;; *-64|*"-64 ") libmagic=64-bit;; *) libmagic=never-match;; esac lt_cv_deplibs_check_method=pass_all ;; # This must be glibc/ELF. linux* | k*bsd*-gnu | kopensolaris*-gnu | gnu*) lt_cv_deplibs_check_method=pass_all ;; netbsd*) if echo __ELF__ | $CC -E - | $GREP __ELF__ > /dev/null; then lt_cv_deplibs_check_method='match_pattern /lib[^/]+(\.so\.[0-9]+\.[0-9]+|_pic\.a)$' else lt_cv_deplibs_check_method='match_pattern /lib[^/]+(\.so|_pic\.a)$' fi ;; newos6*) lt_cv_deplibs_check_method='file_magic ELF [0-9][0-9]*-bit [ML]SB (executable|dynamic lib)' lt_cv_file_magic_cmd=/usr/bin/file lt_cv_file_magic_test_file=/usr/lib/libnls.so ;; *nto* | *qnx*) lt_cv_deplibs_check_method=pass_all ;; openbsd* | bitrig*) if test -z "`echo __ELF__ | $CC -E - | $GREP __ELF__`"; then lt_cv_deplibs_check_method='match_pattern /lib[^/]+(\.so\.[0-9]+\.[0-9]+|\.so|_pic\.a)$' else lt_cv_deplibs_check_method='match_pattern /lib[^/]+(\.so\.[0-9]+\.[0-9]+|_pic\.a)$' fi ;; osf3* | osf4* | osf5*) lt_cv_deplibs_check_method=pass_all ;; rdos*) lt_cv_deplibs_check_method=pass_all ;; solaris*) lt_cv_deplibs_check_method=pass_all ;; sysv5* | sco3.2v5* | sco5v6* | unixware* | OpenUNIX* | sysv4*uw2*) lt_cv_deplibs_check_method=pass_all ;; sysv4 | sysv4.3*) case $host_vendor in motorola) lt_cv_deplibs_check_method='file_magic ELF [0-9][0-9]*-bit [ML]SB (shared object|dynamic lib) M[0-9][0-9]* Version [0-9]' lt_cv_file_magic_test_file=`echo /usr/lib/libc.so*` ;; ncr) lt_cv_deplibs_check_method=pass_all ;; sequent) lt_cv_file_magic_cmd='/bin/file' lt_cv_deplibs_check_method='file_magic ELF [0-9][0-9]*-bit [LM]SB (shared object|dynamic lib )' ;; sni) lt_cv_file_magic_cmd='/bin/file' lt_cv_deplibs_check_method="file_magic ELF [0-9][0-9]*-bit [LM]SB dynamic lib" lt_cv_file_magic_test_file=/lib/libc.so ;; siemens) lt_cv_deplibs_check_method=pass_all ;; pc) lt_cv_deplibs_check_method=pass_all ;; esac ;; tpf*) lt_cv_deplibs_check_method=pass_all ;; os2*) lt_cv_deplibs_check_method=pass_all ;; esac fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_deplibs_check_method" >&5 $as_echo "$lt_cv_deplibs_check_method" >&6; } file_magic_glob= want_nocaseglob=no if test "$build" = "$host"; then case $host_os in mingw* | pw32*) if ( shopt | grep nocaseglob ) >/dev/null 2>&1; then want_nocaseglob=yes else file_magic_glob=`echo aAbBcCdDeEfFgGhHiIjJkKlLmMnNoOpPqQrRsStTuUvVwWxXyYzZ | $SED -e "s/\(..\)/s\/[\1]\/[\1]\/g;/g"` fi ;; esac fi file_magic_cmd=$lt_cv_file_magic_cmd deplibs_check_method=$lt_cv_deplibs_check_method test -z "$deplibs_check_method" && deplibs_check_method=unknown if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}dlltool", so it can be a program name with args. set dummy ${ac_tool_prefix}dlltool; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_DLLTOOL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$DLLTOOL"; then ac_cv_prog_DLLTOOL="$DLLTOOL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_DLLTOOL="${ac_tool_prefix}dlltool" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi DLLTOOL=$ac_cv_prog_DLLTOOL if test -n "$DLLTOOL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $DLLTOOL" >&5 $as_echo "$DLLTOOL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_DLLTOOL"; then ac_ct_DLLTOOL=$DLLTOOL # Extract the first word of "dlltool", so it can be a program name with args. set dummy dlltool; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_DLLTOOL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_DLLTOOL"; then ac_cv_prog_ac_ct_DLLTOOL="$ac_ct_DLLTOOL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_DLLTOOL="dlltool" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_DLLTOOL=$ac_cv_prog_ac_ct_DLLTOOL if test -n "$ac_ct_DLLTOOL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_DLLTOOL" >&5 $as_echo "$ac_ct_DLLTOOL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_DLLTOOL" = x; then DLLTOOL="false" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac DLLTOOL=$ac_ct_DLLTOOL fi else DLLTOOL="$ac_cv_prog_DLLTOOL" fi test -z "$DLLTOOL" && DLLTOOL=dlltool { $as_echo "$as_me:${as_lineno-$LINENO}: checking how to associate runtime and link libraries" >&5 $as_echo_n "checking how to associate runtime and link libraries... " >&6; } if ${lt_cv_sharedlib_from_linklib_cmd+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_sharedlib_from_linklib_cmd='unknown' case $host_os in cygwin* | mingw* | pw32* | cegcc*) # two different shell functions defined in ltmain.sh; # decide which one to use based on capabilities of $DLLTOOL case `$DLLTOOL --help 2>&1` in *--identify-strict*) lt_cv_sharedlib_from_linklib_cmd=func_cygming_dll_for_implib ;; *) lt_cv_sharedlib_from_linklib_cmd=func_cygming_dll_for_implib_fallback ;; esac ;; *) # fallback: assume linklib IS sharedlib lt_cv_sharedlib_from_linklib_cmd=$ECHO ;; esac fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_sharedlib_from_linklib_cmd" >&5 $as_echo "$lt_cv_sharedlib_from_linklib_cmd" >&6; } sharedlib_from_linklib_cmd=$lt_cv_sharedlib_from_linklib_cmd test -z "$sharedlib_from_linklib_cmd" && sharedlib_from_linklib_cmd=$ECHO if test -n "$ac_tool_prefix"; then for ac_prog in ar do # Extract the first word of "$ac_tool_prefix$ac_prog", so it can be a program name with args. set dummy $ac_tool_prefix$ac_prog; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_AR+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$AR"; then ac_cv_prog_AR="$AR" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_AR="$ac_tool_prefix$ac_prog" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi AR=$ac_cv_prog_AR if test -n "$AR"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $AR" >&5 $as_echo "$AR" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi test -n "$AR" && break done fi if test -z "$AR"; then ac_ct_AR=$AR for ac_prog in ar do # Extract the first word of "$ac_prog", so it can be a program name with args. set dummy $ac_prog; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_AR+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_AR"; then ac_cv_prog_ac_ct_AR="$ac_ct_AR" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_AR="$ac_prog" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_AR=$ac_cv_prog_ac_ct_AR if test -n "$ac_ct_AR"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_AR" >&5 $as_echo "$ac_ct_AR" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi test -n "$ac_ct_AR" && break done if test "x$ac_ct_AR" = x; then AR="false" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac AR=$ac_ct_AR fi fi : ${AR=ar} : ${AR_FLAGS=cru} { $as_echo "$as_me:${as_lineno-$LINENO}: checking for archiver @FILE support" >&5 $as_echo_n "checking for archiver @FILE support... " >&6; } if ${lt_cv_ar_at_file+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_ar_at_file=no cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ int main () { ; return 0; } _ACEOF if ac_fn_c_try_compile "$LINENO"; then : echo conftest.$ac_objext > conftest.lst lt_ar_try='$AR $AR_FLAGS libconftest.a @conftest.lst >&5' { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$lt_ar_try\""; } >&5 (eval $lt_ar_try) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; } if test 0 -eq "$ac_status"; then # Ensure the archiver fails upon bogus file names. rm -f conftest.$ac_objext libconftest.a { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$lt_ar_try\""; } >&5 (eval $lt_ar_try) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; } if test 0 -ne "$ac_status"; then lt_cv_ar_at_file=@ fi fi rm -f conftest.* libconftest.a fi rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_ar_at_file" >&5 $as_echo "$lt_cv_ar_at_file" >&6; } if test no = "$lt_cv_ar_at_file"; then archiver_list_spec= else archiver_list_spec=$lt_cv_ar_at_file fi if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}strip", so it can be a program name with args. set dummy ${ac_tool_prefix}strip; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_STRIP+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$STRIP"; then ac_cv_prog_STRIP="$STRIP" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_STRIP="${ac_tool_prefix}strip" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi STRIP=$ac_cv_prog_STRIP if test -n "$STRIP"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $STRIP" >&5 $as_echo "$STRIP" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_STRIP"; then ac_ct_STRIP=$STRIP # Extract the first word of "strip", so it can be a program name with args. set dummy strip; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_STRIP+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_STRIP"; then ac_cv_prog_ac_ct_STRIP="$ac_ct_STRIP" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_STRIP="strip" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_STRIP=$ac_cv_prog_ac_ct_STRIP if test -n "$ac_ct_STRIP"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_STRIP" >&5 $as_echo "$ac_ct_STRIP" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_STRIP" = x; then STRIP=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac STRIP=$ac_ct_STRIP fi else STRIP="$ac_cv_prog_STRIP" fi test -z "$STRIP" && STRIP=: if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}ranlib", so it can be a program name with args. set dummy ${ac_tool_prefix}ranlib; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_RANLIB+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$RANLIB"; then ac_cv_prog_RANLIB="$RANLIB" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_RANLIB="${ac_tool_prefix}ranlib" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi RANLIB=$ac_cv_prog_RANLIB if test -n "$RANLIB"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $RANLIB" >&5 $as_echo "$RANLIB" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_RANLIB"; then ac_ct_RANLIB=$RANLIB # Extract the first word of "ranlib", so it can be a program name with args. set dummy ranlib; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_RANLIB+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_RANLIB"; then ac_cv_prog_ac_ct_RANLIB="$ac_ct_RANLIB" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_RANLIB="ranlib" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_RANLIB=$ac_cv_prog_ac_ct_RANLIB if test -n "$ac_ct_RANLIB"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_RANLIB" >&5 $as_echo "$ac_ct_RANLIB" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_RANLIB" = x; then RANLIB=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac RANLIB=$ac_ct_RANLIB fi else RANLIB="$ac_cv_prog_RANLIB" fi test -z "$RANLIB" && RANLIB=: # Determine commands to create old-style static archives. old_archive_cmds='$AR $AR_FLAGS $oldlib$oldobjs' old_postinstall_cmds='chmod 644 $oldlib' old_postuninstall_cmds= if test -n "$RANLIB"; then case $host_os in bitrig* | openbsd*) old_postinstall_cmds="$old_postinstall_cmds~\$RANLIB -t \$tool_oldlib" ;; *) old_postinstall_cmds="$old_postinstall_cmds~\$RANLIB \$tool_oldlib" ;; esac old_archive_cmds="$old_archive_cmds~\$RANLIB \$tool_oldlib" fi case $host_os in darwin*) lock_old_archive_extraction=yes ;; *) lock_old_archive_extraction=no ;; esac # If no C compiler was specified, use CC. LTCC=${LTCC-"$CC"} # If no C compiler flags were specified, use CFLAGS. LTCFLAGS=${LTCFLAGS-"$CFLAGS"} # Allow CC to be a program name with arguments. compiler=$CC # Check for command to grab the raw symbol name followed by C symbol from nm. { $as_echo "$as_me:${as_lineno-$LINENO}: checking command to parse $NM output from $compiler object" >&5 $as_echo_n "checking command to parse $NM output from $compiler object... " >&6; } if ${lt_cv_sys_global_symbol_pipe+:} false; then : $as_echo_n "(cached) " >&6 else # These are sane defaults that work on at least a few old systems. # [They come from Ultrix. What could be older than Ultrix?!! ;)] # Character class describing NM global symbol codes. symcode='[BCDEGRST]' # Regexp to match symbols that can be accessed directly from C. sympat='\([_A-Za-z][_A-Za-z0-9]*\)' # Define system-specific variables. case $host_os in aix*) symcode='[BCDT]' ;; cygwin* | mingw* | pw32* | cegcc*) symcode='[ABCDGISTW]' ;; hpux*) if test ia64 = "$host_cpu"; then symcode='[ABCDEGRST]' fi ;; irix* | nonstopux*) symcode='[BCDEGRST]' ;; osf*) symcode='[BCDEGQRST]' ;; solaris*) symcode='[BDRT]' ;; sco3.2v5*) symcode='[DT]' ;; sysv4.2uw2*) symcode='[DT]' ;; sysv5* | sco5v6* | unixware* | OpenUNIX*) symcode='[ABDT]' ;; sysv4) symcode='[DFNSTU]' ;; esac # If we're using GNU nm, then use its standard symbol codes. case `$NM -V 2>&1` in *GNU* | *'with BFD'*) symcode='[ABCDGIRSTW]' ;; esac if test "$lt_cv_nm_interface" = "MS dumpbin"; then # Gets list of data symbols to import. lt_cv_sys_global_symbol_to_import="sed -n -e 's/^I .* \(.*\)$/\1/p'" # Adjust the below global symbol transforms to fixup imported variables. lt_cdecl_hook=" -e 's/^I .* \(.*\)$/extern __declspec(dllimport) char \1;/p'" lt_c_name_hook=" -e 's/^I .* \(.*\)$/ {\"\1\", (void *) 0},/p'" lt_c_name_lib_hook="\ -e 's/^I .* \(lib.*\)$/ {\"\1\", (void *) 0},/p'\ -e 's/^I .* \(.*\)$/ {\"lib\1\", (void *) 0},/p'" else # Disable hooks by default. lt_cv_sys_global_symbol_to_import= lt_cdecl_hook= lt_c_name_hook= lt_c_name_lib_hook= fi # Transform an extracted symbol line into a proper C declaration. # Some systems (esp. on ia64) link data and code symbols differently, # so use this general approach. lt_cv_sys_global_symbol_to_cdecl="sed -n"\ $lt_cdecl_hook\ " -e 's/^T .* \(.*\)$/extern int \1();/p'"\ " -e 's/^$symcode$symcode* .* \(.*\)$/extern char \1;/p'" # Transform an extracted symbol line into symbol name and symbol address lt_cv_sys_global_symbol_to_c_name_address="sed -n"\ $lt_c_name_hook\ " -e 's/^: \(.*\) .*$/ {\"\1\", (void *) 0},/p'"\ " -e 's/^$symcode$symcode* .* \(.*\)$/ {\"\1\", (void *) \&\1},/p'" # Transform an extracted symbol line into symbol name with lib prefix and # symbol address. lt_cv_sys_global_symbol_to_c_name_address_lib_prefix="sed -n"\ $lt_c_name_lib_hook\ " -e 's/^: \(.*\) .*$/ {\"\1\", (void *) 0},/p'"\ " -e 's/^$symcode$symcode* .* \(lib.*\)$/ {\"\1\", (void *) \&\1},/p'"\ " -e 's/^$symcode$symcode* .* \(.*\)$/ {\"lib\1\", (void *) \&\1},/p'" # Handle CRLF in mingw tool chain opt_cr= case $build_os in mingw*) opt_cr=`$ECHO 'x\{0,1\}' | tr x '\015'` # option cr in regexp ;; esac # Try without a prefix underscore, then with it. for ac_symprfx in "" "_"; do # Transform symcode, sympat, and symprfx into a raw symbol and a C symbol. symxfrm="\\1 $ac_symprfx\\2 \\2" # Write the raw and C identifiers. if test "$lt_cv_nm_interface" = "MS dumpbin"; then # Fake it for dumpbin and say T for any non-static function, # D for any global variable and I for any imported variable. # Also find C++ and __fastcall symbols from MSVC++, # which start with @ or ?. lt_cv_sys_global_symbol_pipe="$AWK '"\ " {last_section=section; section=\$ 3};"\ " /^COFF SYMBOL TABLE/{for(i in hide) delete hide[i]};"\ " /Section length .*#relocs.*(pick any)/{hide[last_section]=1};"\ " /^ *Symbol name *: /{split(\$ 0,sn,\":\"); si=substr(sn[2],2)};"\ " /^ *Type *: code/{print \"T\",si,substr(si,length(prfx))};"\ " /^ *Type *: data/{print \"I\",si,substr(si,length(prfx))};"\ " \$ 0!~/External *\|/{next};"\ " / 0+ UNDEF /{next}; / UNDEF \([^|]\)*()/{next};"\ " {if(hide[section]) next};"\ " {f=\"D\"}; \$ 0~/\(\).*\|/{f=\"T\"};"\ " {split(\$ 0,a,/\||\r/); split(a[2],s)};"\ " s[1]~/^[@?]/{print f,s[1],s[1]; next};"\ " s[1]~prfx {split(s[1],t,\"@\"); print f,t[1],substr(t[1],length(prfx))}"\ " ' prfx=^$ac_symprfx" else lt_cv_sys_global_symbol_pipe="sed -n -e 's/^.*[ ]\($symcode$symcode*\)[ ][ ]*$ac_symprfx$sympat$opt_cr$/$symxfrm/p'" fi lt_cv_sys_global_symbol_pipe="$lt_cv_sys_global_symbol_pipe | sed '/ __gnu_lto/d'" # Check to see that the pipe works correctly. pipe_works=no rm -f conftest* cat > conftest.$ac_ext <<_LT_EOF #ifdef __cplusplus extern "C" { #endif char nm_test_var; void nm_test_func(void); void nm_test_func(void){} #ifdef __cplusplus } #endif int main(){nm_test_var='a';nm_test_func();return(0);} _LT_EOF if { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$ac_compile\""; } >&5 (eval $ac_compile) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; }; then # Now try to grab the symbols. nlist=conftest.nm if { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$NM conftest.$ac_objext \| "$lt_cv_sys_global_symbol_pipe" \> $nlist\""; } >&5 (eval $NM conftest.$ac_objext \| "$lt_cv_sys_global_symbol_pipe" \> $nlist) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; } && test -s "$nlist"; then # Try sorting and uniquifying the output. if sort "$nlist" | uniq > "$nlist"T; then mv -f "$nlist"T "$nlist" else rm -f "$nlist"T fi # Make sure that we snagged all the symbols we need. if $GREP ' nm_test_var$' "$nlist" >/dev/null; then if $GREP ' nm_test_func$' "$nlist" >/dev/null; then cat <<_LT_EOF > conftest.$ac_ext /* Keep this code in sync between libtool.m4, ltmain, lt_system.h, and tests. */ #if defined _WIN32 || defined __CYGWIN__ || defined _WIN32_WCE /* DATA imports from DLLs on WIN32 can't be const, because runtime relocations are performed -- see ld's documentation on pseudo-relocs. */ # define LT_DLSYM_CONST #elif defined __osf__ /* This system does not cope well with relocations in const data. */ # define LT_DLSYM_CONST #else # define LT_DLSYM_CONST const #endif #ifdef __cplusplus extern "C" { #endif _LT_EOF # Now generate the symbol file. eval "$lt_cv_sys_global_symbol_to_cdecl"' < "$nlist" | $GREP -v main >> conftest.$ac_ext' cat <<_LT_EOF >> conftest.$ac_ext /* The mapping between symbol names and symbols. */ LT_DLSYM_CONST struct { const char *name; void *address; } lt__PROGRAM__LTX_preloaded_symbols[] = { { "@PROGRAM@", (void *) 0 }, _LT_EOF $SED "s/^$symcode$symcode* .* \(.*\)$/ {\"\1\", (void *) \&\1},/" < "$nlist" | $GREP -v main >> conftest.$ac_ext cat <<\_LT_EOF >> conftest.$ac_ext {0, (void *) 0} }; /* This works around a problem in FreeBSD linker */ #ifdef FREEBSD_WORKAROUND static const void *lt_preloaded_setup() { return lt__PROGRAM__LTX_preloaded_symbols; } #endif #ifdef __cplusplus } #endif _LT_EOF # Now try linking the two files. mv conftest.$ac_objext conftstm.$ac_objext lt_globsym_save_LIBS=$LIBS lt_globsym_save_CFLAGS=$CFLAGS LIBS=conftstm.$ac_objext CFLAGS="$CFLAGS$lt_prog_compiler_no_builtin_flag" if { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$ac_link\""; } >&5 (eval $ac_link) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; } && test -s conftest$ac_exeext; then pipe_works=yes fi LIBS=$lt_globsym_save_LIBS CFLAGS=$lt_globsym_save_CFLAGS else echo "cannot find nm_test_func in $nlist" >&5 fi else echo "cannot find nm_test_var in $nlist" >&5 fi else echo "cannot run $lt_cv_sys_global_symbol_pipe" >&5 fi else echo "$progname: failed program was:" >&5 cat conftest.$ac_ext >&5 fi rm -rf conftest* conftst* # Do not use the global_symbol_pipe unless it works. if test yes = "$pipe_works"; then break else lt_cv_sys_global_symbol_pipe= fi done fi if test -z "$lt_cv_sys_global_symbol_pipe"; then lt_cv_sys_global_symbol_to_cdecl= fi if test -z "$lt_cv_sys_global_symbol_pipe$lt_cv_sys_global_symbol_to_cdecl"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: failed" >&5 $as_echo "failed" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: ok" >&5 $as_echo "ok" >&6; } fi # Response file support. if test "$lt_cv_nm_interface" = "MS dumpbin"; then nm_file_list_spec='@' elif $NM --help 2>/dev/null | grep '[@]FILE' >/dev/null; then nm_file_list_spec='@' fi { $as_echo "$as_me:${as_lineno-$LINENO}: checking for sysroot" >&5 $as_echo_n "checking for sysroot... " >&6; } # Check whether --with-sysroot was given. if test "${with_sysroot+set}" = set; then : withval=$with_sysroot; else with_sysroot=no fi lt_sysroot= case $with_sysroot in #( yes) if test yes = "$GCC"; then lt_sysroot=`$CC --print-sysroot 2>/dev/null` fi ;; #( /*) lt_sysroot=`echo "$with_sysroot" | sed -e "$sed_quote_subst"` ;; #( no|'') ;; #( *) { $as_echo "$as_me:${as_lineno-$LINENO}: result: $with_sysroot" >&5 $as_echo "$with_sysroot" >&6; } as_fn_error $? "The sysroot must be an absolute path." "$LINENO" 5 ;; esac { $as_echo "$as_me:${as_lineno-$LINENO}: result: ${lt_sysroot:-no}" >&5 $as_echo "${lt_sysroot:-no}" >&6; } { $as_echo "$as_me:${as_lineno-$LINENO}: checking for a working dd" >&5 $as_echo_n "checking for a working dd... " >&6; } if ${ac_cv_path_lt_DD+:} false; then : $as_echo_n "(cached) " >&6 else printf 0123456789abcdef0123456789abcdef >conftest.i cat conftest.i conftest.i >conftest2.i : ${lt_DD:=$DD} if test -z "$lt_DD"; then ac_path_lt_DD_found=false # Loop through the user's path and test for each of PROGNAME-LIST as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_prog in dd; do for ac_exec_ext in '' $ac_executable_extensions; do ac_path_lt_DD="$as_dir/$ac_prog$ac_exec_ext" as_fn_executable_p "$ac_path_lt_DD" || continue if "$ac_path_lt_DD" bs=32 count=1 conftest.out 2>/dev/null; then cmp -s conftest.i conftest.out \ && ac_cv_path_lt_DD="$ac_path_lt_DD" ac_path_lt_DD_found=: fi $ac_path_lt_DD_found && break 3 done done done IFS=$as_save_IFS if test -z "$ac_cv_path_lt_DD"; then : fi else ac_cv_path_lt_DD=$lt_DD fi rm -f conftest.i conftest2.i conftest.out fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_cv_path_lt_DD" >&5 $as_echo "$ac_cv_path_lt_DD" >&6; } { $as_echo "$as_me:${as_lineno-$LINENO}: checking how to truncate binary pipes" >&5 $as_echo_n "checking how to truncate binary pipes... " >&6; } if ${lt_cv_truncate_bin+:} false; then : $as_echo_n "(cached) " >&6 else printf 0123456789abcdef0123456789abcdef >conftest.i cat conftest.i conftest.i >conftest2.i lt_cv_truncate_bin= if "$ac_cv_path_lt_DD" bs=32 count=1 conftest.out 2>/dev/null; then cmp -s conftest.i conftest.out \ && lt_cv_truncate_bin="$ac_cv_path_lt_DD bs=4096 count=1" fi rm -f conftest.i conftest2.i conftest.out test -z "$lt_cv_truncate_bin" && lt_cv_truncate_bin="$SED -e 4q" fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_truncate_bin" >&5 $as_echo "$lt_cv_truncate_bin" >&6; } # Calculate cc_basename. Skip known compiler wrappers and cross-prefix. func_cc_basename () { for cc_temp in $*""; do case $cc_temp in compile | *[\\/]compile | ccache | *[\\/]ccache ) ;; distcc | *[\\/]distcc | purify | *[\\/]purify ) ;; \-*) ;; *) break;; esac done func_cc_basename_result=`$ECHO "$cc_temp" | $SED "s%.*/%%; s%^$host_alias-%%"` } # Check whether --enable-libtool-lock was given. if test "${enable_libtool_lock+set}" = set; then : enableval=$enable_libtool_lock; fi test no = "$enable_libtool_lock" || enable_libtool_lock=yes # Some flags need to be propagated to the compiler or linker for good # libtool support. case $host in ia64-*-hpux*) # Find out what ABI is being produced by ac_compile, and set mode # options accordingly. echo 'int i;' > conftest.$ac_ext if { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$ac_compile\""; } >&5 (eval $ac_compile) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; }; then case `/usr/bin/file conftest.$ac_objext` in *ELF-32*) HPUX_IA64_MODE=32 ;; *ELF-64*) HPUX_IA64_MODE=64 ;; esac fi rm -rf conftest* ;; *-*-irix6*) # Find out what ABI is being produced by ac_compile, and set linker # options accordingly. echo '#line '$LINENO' "configure"' > conftest.$ac_ext if { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$ac_compile\""; } >&5 (eval $ac_compile) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; }; then if test yes = "$lt_cv_prog_gnu_ld"; then case `/usr/bin/file conftest.$ac_objext` in *32-bit*) LD="${LD-ld} -melf32bsmip" ;; *N32*) LD="${LD-ld} -melf32bmipn32" ;; *64-bit*) LD="${LD-ld} -melf64bmip" ;; esac else case `/usr/bin/file conftest.$ac_objext` in *32-bit*) LD="${LD-ld} -32" ;; *N32*) LD="${LD-ld} -n32" ;; *64-bit*) LD="${LD-ld} -64" ;; esac fi fi rm -rf conftest* ;; mips64*-*linux*) # Find out what ABI is being produced by ac_compile, and set linker # options accordingly. echo '#line '$LINENO' "configure"' > conftest.$ac_ext if { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$ac_compile\""; } >&5 (eval $ac_compile) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; }; then emul=elf case `/usr/bin/file conftest.$ac_objext` in *32-bit*) emul="${emul}32" ;; *64-bit*) emul="${emul}64" ;; esac case `/usr/bin/file conftest.$ac_objext` in *MSB*) emul="${emul}btsmip" ;; *LSB*) emul="${emul}ltsmip" ;; esac case `/usr/bin/file conftest.$ac_objext` in *N32*) emul="${emul}n32" ;; esac LD="${LD-ld} -m $emul" fi rm -rf conftest* ;; x86_64-*kfreebsd*-gnu|x86_64-*linux*|powerpc*-*linux*| \ s390*-*linux*|s390*-*tpf*|sparc*-*linux*) # Find out what ABI is being produced by ac_compile, and set linker # options accordingly. Note that the listed cases only cover the # situations where additional linker options are needed (such as when # doing 32-bit compilation for a host where ld defaults to 64-bit, or # vice versa); the common cases where no linker options are needed do # not appear in the list. echo 'int i;' > conftest.$ac_ext if { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$ac_compile\""; } >&5 (eval $ac_compile) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; }; then case `/usr/bin/file conftest.o` in *32-bit*) case $host in x86_64-*kfreebsd*-gnu) LD="${LD-ld} -m elf_i386_fbsd" ;; x86_64-*linux*) case `/usr/bin/file conftest.o` in *x86-64*) LD="${LD-ld} -m elf32_x86_64" ;; *) LD="${LD-ld} -m elf_i386" ;; esac ;; powerpc64le-*linux*) LD="${LD-ld} -m elf32lppclinux" ;; powerpc64-*linux*) LD="${LD-ld} -m elf32ppclinux" ;; s390x-*linux*) LD="${LD-ld} -m elf_s390" ;; sparc64-*linux*) LD="${LD-ld} -m elf32_sparc" ;; esac ;; *64-bit*) case $host in x86_64-*kfreebsd*-gnu) LD="${LD-ld} -m elf_x86_64_fbsd" ;; x86_64-*linux*) LD="${LD-ld} -m elf_x86_64" ;; powerpcle-*linux*) LD="${LD-ld} -m elf64lppc" ;; powerpc-*linux*) LD="${LD-ld} -m elf64ppc" ;; s390*-*linux*|s390*-*tpf*) LD="${LD-ld} -m elf64_s390" ;; sparc*-*linux*) LD="${LD-ld} -m elf64_sparc" ;; esac ;; esac fi rm -rf conftest* ;; *-*-sco3.2v5*) # On SCO OpenServer 5, we need -belf to get full-featured binaries. SAVE_CFLAGS=$CFLAGS CFLAGS="$CFLAGS -belf" { $as_echo "$as_me:${as_lineno-$LINENO}: checking whether the C compiler needs -belf" >&5 $as_echo_n "checking whether the C compiler needs -belf... " >&6; } if ${lt_cv_cc_needs_belf+:} false; then : $as_echo_n "(cached) " >&6 else ac_ext=c ac_cpp='$CPP $CPPFLAGS' ac_compile='$CC -c $CFLAGS $CPPFLAGS conftest.$ac_ext >&5' ac_link='$CC -o conftest$ac_exeext $CFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5' ac_compiler_gnu=$ac_cv_c_compiler_gnu cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ int main () { ; return 0; } _ACEOF if ac_fn_c_try_link "$LINENO"; then : lt_cv_cc_needs_belf=yes else lt_cv_cc_needs_belf=no fi rm -f core conftest.err conftest.$ac_objext \ conftest$ac_exeext conftest.$ac_ext ac_ext=c ac_cpp='$CPP $CPPFLAGS' ac_compile='$CC -c $CFLAGS $CPPFLAGS conftest.$ac_ext >&5' ac_link='$CC -o conftest$ac_exeext $CFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5' ac_compiler_gnu=$ac_cv_c_compiler_gnu fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_cc_needs_belf" >&5 $as_echo "$lt_cv_cc_needs_belf" >&6; } if test yes != "$lt_cv_cc_needs_belf"; then # this is probably gcc 2.8.0, egcs 1.0 or newer; no need for -belf CFLAGS=$SAVE_CFLAGS fi ;; *-*solaris*) # Find out what ABI is being produced by ac_compile, and set linker # options accordingly. echo 'int i;' > conftest.$ac_ext if { { eval echo "\"\$as_me\":${as_lineno-$LINENO}: \"$ac_compile\""; } >&5 (eval $ac_compile) 2>&5 ac_status=$? $as_echo "$as_me:${as_lineno-$LINENO}: \$? = $ac_status" >&5 test $ac_status = 0; }; then case `/usr/bin/file conftest.o` in *64-bit*) case $lt_cv_prog_gnu_ld in yes*) case $host in i?86-*-solaris*|x86_64-*-solaris*) LD="${LD-ld} -m elf_x86_64" ;; sparc*-*-solaris*) LD="${LD-ld} -m elf64_sparc" ;; esac # GNU ld 2.21 introduced _sol2 emulations. Use them if available. if ${LD-ld} -V | grep _sol2 >/dev/null 2>&1; then LD=${LD-ld}_sol2 fi ;; *) if ${LD-ld} -64 -r -o conftest2.o conftest.o >/dev/null 2>&1; then LD="${LD-ld} -64" fi ;; esac ;; esac fi rm -rf conftest* ;; esac need_locks=$enable_libtool_lock if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}mt", so it can be a program name with args. set dummy ${ac_tool_prefix}mt; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_MANIFEST_TOOL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$MANIFEST_TOOL"; then ac_cv_prog_MANIFEST_TOOL="$MANIFEST_TOOL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_MANIFEST_TOOL="${ac_tool_prefix}mt" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi MANIFEST_TOOL=$ac_cv_prog_MANIFEST_TOOL if test -n "$MANIFEST_TOOL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $MANIFEST_TOOL" >&5 $as_echo "$MANIFEST_TOOL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_MANIFEST_TOOL"; then ac_ct_MANIFEST_TOOL=$MANIFEST_TOOL # Extract the first word of "mt", so it can be a program name with args. set dummy mt; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_MANIFEST_TOOL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_MANIFEST_TOOL"; then ac_cv_prog_ac_ct_MANIFEST_TOOL="$ac_ct_MANIFEST_TOOL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_MANIFEST_TOOL="mt" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_MANIFEST_TOOL=$ac_cv_prog_ac_ct_MANIFEST_TOOL if test -n "$ac_ct_MANIFEST_TOOL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_MANIFEST_TOOL" >&5 $as_echo "$ac_ct_MANIFEST_TOOL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_MANIFEST_TOOL" = x; then MANIFEST_TOOL=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac MANIFEST_TOOL=$ac_ct_MANIFEST_TOOL fi else MANIFEST_TOOL="$ac_cv_prog_MANIFEST_TOOL" fi test -z "$MANIFEST_TOOL" && MANIFEST_TOOL=mt { $as_echo "$as_me:${as_lineno-$LINENO}: checking if $MANIFEST_TOOL is a manifest tool" >&5 $as_echo_n "checking if $MANIFEST_TOOL is a manifest tool... " >&6; } if ${lt_cv_path_mainfest_tool+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_path_mainfest_tool=no echo "$as_me:$LINENO: $MANIFEST_TOOL '-?'" >&5 $MANIFEST_TOOL '-?' 2>conftest.err > conftest.out cat conftest.err >&5 if $GREP 'Manifest Tool' conftest.out > /dev/null; then lt_cv_path_mainfest_tool=yes fi rm -f conftest* fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_path_mainfest_tool" >&5 $as_echo "$lt_cv_path_mainfest_tool" >&6; } if test yes != "$lt_cv_path_mainfest_tool"; then MANIFEST_TOOL=: fi case $host_os in rhapsody* | darwin*) if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}dsymutil", so it can be a program name with args. set dummy ${ac_tool_prefix}dsymutil; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_DSYMUTIL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$DSYMUTIL"; then ac_cv_prog_DSYMUTIL="$DSYMUTIL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_DSYMUTIL="${ac_tool_prefix}dsymutil" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi DSYMUTIL=$ac_cv_prog_DSYMUTIL if test -n "$DSYMUTIL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $DSYMUTIL" >&5 $as_echo "$DSYMUTIL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_DSYMUTIL"; then ac_ct_DSYMUTIL=$DSYMUTIL # Extract the first word of "dsymutil", so it can be a program name with args. set dummy dsymutil; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_DSYMUTIL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_DSYMUTIL"; then ac_cv_prog_ac_ct_DSYMUTIL="$ac_ct_DSYMUTIL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_DSYMUTIL="dsymutil" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_DSYMUTIL=$ac_cv_prog_ac_ct_DSYMUTIL if test -n "$ac_ct_DSYMUTIL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_DSYMUTIL" >&5 $as_echo "$ac_ct_DSYMUTIL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_DSYMUTIL" = x; then DSYMUTIL=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac DSYMUTIL=$ac_ct_DSYMUTIL fi else DSYMUTIL="$ac_cv_prog_DSYMUTIL" fi if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}nmedit", so it can be a program name with args. set dummy ${ac_tool_prefix}nmedit; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_NMEDIT+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$NMEDIT"; then ac_cv_prog_NMEDIT="$NMEDIT" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_NMEDIT="${ac_tool_prefix}nmedit" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi NMEDIT=$ac_cv_prog_NMEDIT if test -n "$NMEDIT"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $NMEDIT" >&5 $as_echo "$NMEDIT" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_NMEDIT"; then ac_ct_NMEDIT=$NMEDIT # Extract the first word of "nmedit", so it can be a program name with args. set dummy nmedit; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_NMEDIT+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_NMEDIT"; then ac_cv_prog_ac_ct_NMEDIT="$ac_ct_NMEDIT" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_NMEDIT="nmedit" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_NMEDIT=$ac_cv_prog_ac_ct_NMEDIT if test -n "$ac_ct_NMEDIT"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_NMEDIT" >&5 $as_echo "$ac_ct_NMEDIT" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_NMEDIT" = x; then NMEDIT=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac NMEDIT=$ac_ct_NMEDIT fi else NMEDIT="$ac_cv_prog_NMEDIT" fi if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}lipo", so it can be a program name with args. set dummy ${ac_tool_prefix}lipo; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_LIPO+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$LIPO"; then ac_cv_prog_LIPO="$LIPO" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_LIPO="${ac_tool_prefix}lipo" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi LIPO=$ac_cv_prog_LIPO if test -n "$LIPO"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $LIPO" >&5 $as_echo "$LIPO" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_LIPO"; then ac_ct_LIPO=$LIPO # Extract the first word of "lipo", so it can be a program name with args. set dummy lipo; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_LIPO+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_LIPO"; then ac_cv_prog_ac_ct_LIPO="$ac_ct_LIPO" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_LIPO="lipo" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_LIPO=$ac_cv_prog_ac_ct_LIPO if test -n "$ac_ct_LIPO"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_LIPO" >&5 $as_echo "$ac_ct_LIPO" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_LIPO" = x; then LIPO=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac LIPO=$ac_ct_LIPO fi else LIPO="$ac_cv_prog_LIPO" fi if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}otool", so it can be a program name with args. set dummy ${ac_tool_prefix}otool; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_OTOOL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$OTOOL"; then ac_cv_prog_OTOOL="$OTOOL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_OTOOL="${ac_tool_prefix}otool" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi OTOOL=$ac_cv_prog_OTOOL if test -n "$OTOOL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $OTOOL" >&5 $as_echo "$OTOOL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_OTOOL"; then ac_ct_OTOOL=$OTOOL # Extract the first word of "otool", so it can be a program name with args. set dummy otool; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_OTOOL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_OTOOL"; then ac_cv_prog_ac_ct_OTOOL="$ac_ct_OTOOL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_OTOOL="otool" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_OTOOL=$ac_cv_prog_ac_ct_OTOOL if test -n "$ac_ct_OTOOL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_OTOOL" >&5 $as_echo "$ac_ct_OTOOL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_OTOOL" = x; then OTOOL=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac OTOOL=$ac_ct_OTOOL fi else OTOOL="$ac_cv_prog_OTOOL" fi if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}otool64", so it can be a program name with args. set dummy ${ac_tool_prefix}otool64; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_OTOOL64+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$OTOOL64"; then ac_cv_prog_OTOOL64="$OTOOL64" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_OTOOL64="${ac_tool_prefix}otool64" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi OTOOL64=$ac_cv_prog_OTOOL64 if test -n "$OTOOL64"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $OTOOL64" >&5 $as_echo "$OTOOL64" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_OTOOL64"; then ac_ct_OTOOL64=$OTOOL64 # Extract the first word of "otool64", so it can be a program name with args. set dummy otool64; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_OTOOL64+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_OTOOL64"; then ac_cv_prog_ac_ct_OTOOL64="$ac_ct_OTOOL64" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_OTOOL64="otool64" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_OTOOL64=$ac_cv_prog_ac_ct_OTOOL64 if test -n "$ac_ct_OTOOL64"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_OTOOL64" >&5 $as_echo "$ac_ct_OTOOL64" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_OTOOL64" = x; then OTOOL64=":" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac OTOOL64=$ac_ct_OTOOL64 fi else OTOOL64="$ac_cv_prog_OTOOL64" fi { $as_echo "$as_me:${as_lineno-$LINENO}: checking for -single_module linker flag" >&5 $as_echo_n "checking for -single_module linker flag... " >&6; } if ${lt_cv_apple_cc_single_mod+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_apple_cc_single_mod=no if test -z "$LT_MULTI_MODULE"; then # By default we will add the -single_module flag. You can override # by either setting the environment variable LT_MULTI_MODULE # non-empty at configure time, or by adding -multi_module to the # link flags. rm -rf libconftest.dylib* echo "int foo(void){return 1;}" > conftest.c echo "$LTCC $LTCFLAGS $LDFLAGS -o libconftest.dylib \ -dynamiclib -Wl,-single_module conftest.c" >&5 $LTCC $LTCFLAGS $LDFLAGS -o libconftest.dylib \ -dynamiclib -Wl,-single_module conftest.c 2>conftest.err _lt_result=$? # If there is a non-empty error log, and "single_module" # appears in it, assume the flag caused a linker warning if test -s conftest.err && $GREP single_module conftest.err; then cat conftest.err >&5 # Otherwise, if the output was created with a 0 exit code from # the compiler, it worked. elif test -f libconftest.dylib && test 0 = "$_lt_result"; then lt_cv_apple_cc_single_mod=yes else cat conftest.err >&5 fi rm -rf libconftest.dylib* rm -f conftest.* fi fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_apple_cc_single_mod" >&5 $as_echo "$lt_cv_apple_cc_single_mod" >&6; } { $as_echo "$as_me:${as_lineno-$LINENO}: checking for -exported_symbols_list linker flag" >&5 $as_echo_n "checking for -exported_symbols_list linker flag... " >&6; } if ${lt_cv_ld_exported_symbols_list+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_ld_exported_symbols_list=no save_LDFLAGS=$LDFLAGS echo "_main" > conftest.sym LDFLAGS="$LDFLAGS -Wl,-exported_symbols_list,conftest.sym" cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ int main () { ; return 0; } _ACEOF if ac_fn_c_try_link "$LINENO"; then : lt_cv_ld_exported_symbols_list=yes else lt_cv_ld_exported_symbols_list=no fi rm -f core conftest.err conftest.$ac_objext \ conftest$ac_exeext conftest.$ac_ext LDFLAGS=$save_LDFLAGS fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_ld_exported_symbols_list" >&5 $as_echo "$lt_cv_ld_exported_symbols_list" >&6; } { $as_echo "$as_me:${as_lineno-$LINENO}: checking for -force_load linker flag" >&5 $as_echo_n "checking for -force_load linker flag... " >&6; } if ${lt_cv_ld_force_load+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_ld_force_load=no cat > conftest.c << _LT_EOF int forced_loaded() { return 2;} _LT_EOF echo "$LTCC $LTCFLAGS -c -o conftest.o conftest.c" >&5 $LTCC $LTCFLAGS -c -o conftest.o conftest.c 2>&5 echo "$AR cru libconftest.a conftest.o" >&5 $AR cru libconftest.a conftest.o 2>&5 echo "$RANLIB libconftest.a" >&5 $RANLIB libconftest.a 2>&5 cat > conftest.c << _LT_EOF int main() { return 0;} _LT_EOF echo "$LTCC $LTCFLAGS $LDFLAGS -o conftest conftest.c -Wl,-force_load,./libconftest.a" >&5 $LTCC $LTCFLAGS $LDFLAGS -o conftest conftest.c -Wl,-force_load,./libconftest.a 2>conftest.err _lt_result=$? if test -s conftest.err && $GREP force_load conftest.err; then cat conftest.err >&5 elif test -f conftest && test 0 = "$_lt_result" && $GREP forced_load conftest >/dev/null 2>&1; then lt_cv_ld_force_load=yes else cat conftest.err >&5 fi rm -f conftest.err libconftest.a conftest conftest.c rm -rf conftest.dSYM fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_ld_force_load" >&5 $as_echo "$lt_cv_ld_force_load" >&6; } case $host_os in rhapsody* | darwin1.[012]) _lt_dar_allow_undefined='$wl-undefined ${wl}suppress' ;; darwin1.*) _lt_dar_allow_undefined='$wl-flat_namespace $wl-undefined ${wl}suppress' ;; darwin*) # darwin 5.x on # if running on 10.5 or later, the deployment target defaults # to the OS version, if on x86, and 10.4, the deployment # target defaults to 10.4. Don't you love it? case ${MACOSX_DEPLOYMENT_TARGET-10.0},$host in 10.0,*86*-darwin8*|10.0,*-darwin[91]*) _lt_dar_allow_undefined='$wl-undefined ${wl}dynamic_lookup' ;; 10.[012][,.]*) _lt_dar_allow_undefined='$wl-flat_namespace $wl-undefined ${wl}suppress' ;; 10.*) _lt_dar_allow_undefined='$wl-undefined ${wl}dynamic_lookup' ;; esac ;; esac if test yes = "$lt_cv_apple_cc_single_mod"; then _lt_dar_single_mod='$single_module' fi if test yes = "$lt_cv_ld_exported_symbols_list"; then _lt_dar_export_syms=' $wl-exported_symbols_list,$output_objdir/$libname-symbols.expsym' else _lt_dar_export_syms='~$NMEDIT -s $output_objdir/$libname-symbols.expsym $lib' fi if test : != "$DSYMUTIL" && test no = "$lt_cv_ld_force_load"; then _lt_dsymutil='~$DSYMUTIL $lib || :' else _lt_dsymutil= fi ;; esac # func_munge_path_list VARIABLE PATH # ----------------------------------- # VARIABLE is name of variable containing _space_ separated list of # directories to be munged by the contents of PATH, which is string # having a format: # "DIR[:DIR]:" # string "DIR[ DIR]" will be prepended to VARIABLE # ":DIR[:DIR]" # string "DIR[ DIR]" will be appended to VARIABLE # "DIRP[:DIRP]::[DIRA:]DIRA" # string "DIRP[ DIRP]" will be prepended to VARIABLE and string # "DIRA[ DIRA]" will be appended to VARIABLE # "DIR[:DIR]" # VARIABLE will be replaced by "DIR[ DIR]" func_munge_path_list () { case x$2 in x) ;; *:) eval $1=\"`$ECHO $2 | $SED 's/:/ /g'` \$$1\" ;; x:*) eval $1=\"\$$1 `$ECHO $2 | $SED 's/:/ /g'`\" ;; *::*) eval $1=\"\$$1\ `$ECHO $2 | $SED -e 's/.*:://' -e 's/:/ /g'`\" eval $1=\"`$ECHO $2 | $SED -e 's/::.*//' -e 's/:/ /g'`\ \$$1\" ;; *) eval $1=\"`$ECHO $2 | $SED 's/:/ /g'`\" ;; esac } ac_ext=c ac_cpp='$CPP $CPPFLAGS' ac_compile='$CC -c $CFLAGS $CPPFLAGS conftest.$ac_ext >&5' ac_link='$CC -o conftest$ac_exeext $CFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5' ac_compiler_gnu=$ac_cv_c_compiler_gnu { $as_echo "$as_me:${as_lineno-$LINENO}: checking how to run the C preprocessor" >&5 $as_echo_n "checking how to run the C preprocessor... " >&6; } # On Suns, sometimes $CPP names a directory. if test -n "$CPP" && test -d "$CPP"; then CPP= fi if test -z "$CPP"; then if ${ac_cv_prog_CPP+:} false; then : $as_echo_n "(cached) " >&6 else # Double quotes because CPP needs to be expanded for CPP in "$CC -E" "$CC -E -traditional-cpp" "/lib/cpp" do ac_preproc_ok=false for ac_c_preproc_warn_flag in '' yes do # Use a header file that comes with gcc, so configuring glibc # with a fresh cross-compiler works. # Prefer to if __STDC__ is defined, since # exists even on freestanding compilers. # On the NeXT, cc -E runs the code through the compiler's parser, # not just through cpp. "Syntax error" is here to catch this case. cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ #ifdef __STDC__ # include #else # include #endif Syntax error _ACEOF if ac_fn_c_try_cpp "$LINENO"; then : else # Broken: fails on valid input. continue fi rm -f conftest.err conftest.i conftest.$ac_ext # OK, works on sane cases. Now check whether nonexistent headers # can be detected and how. cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ #include _ACEOF if ac_fn_c_try_cpp "$LINENO"; then : # Broken: success on invalid input. continue else # Passes both tests. ac_preproc_ok=: break fi rm -f conftest.err conftest.i conftest.$ac_ext done # Because of `break', _AC_PREPROC_IFELSE's cleaning code was skipped. rm -f conftest.i conftest.err conftest.$ac_ext if $ac_preproc_ok; then : break fi done ac_cv_prog_CPP=$CPP fi CPP=$ac_cv_prog_CPP else ac_cv_prog_CPP=$CPP fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $CPP" >&5 $as_echo "$CPP" >&6; } ac_preproc_ok=false for ac_c_preproc_warn_flag in '' yes do # Use a header file that comes with gcc, so configuring glibc # with a fresh cross-compiler works. # Prefer to if __STDC__ is defined, since # exists even on freestanding compilers. # On the NeXT, cc -E runs the code through the compiler's parser, # not just through cpp. "Syntax error" is here to catch this case. cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ #ifdef __STDC__ # include #else # include #endif Syntax error _ACEOF if ac_fn_c_try_cpp "$LINENO"; then : else # Broken: fails on valid input. continue fi rm -f conftest.err conftest.i conftest.$ac_ext # OK, works on sane cases. Now check whether nonexistent headers # can be detected and how. cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ #include _ACEOF if ac_fn_c_try_cpp "$LINENO"; then : # Broken: success on invalid input. continue else # Passes both tests. ac_preproc_ok=: break fi rm -f conftest.err conftest.i conftest.$ac_ext done # Because of `break', _AC_PREPROC_IFELSE's cleaning code was skipped. rm -f conftest.i conftest.err conftest.$ac_ext if $ac_preproc_ok; then : else { { $as_echo "$as_me:${as_lineno-$LINENO}: error: in \`$ac_pwd':" >&5 $as_echo "$as_me: error: in \`$ac_pwd':" >&2;} as_fn_error $? "C preprocessor \"$CPP\" fails sanity check See \`config.log' for more details" "$LINENO" 5; } fi ac_ext=c ac_cpp='$CPP $CPPFLAGS' ac_compile='$CC -c $CFLAGS $CPPFLAGS conftest.$ac_ext >&5' ac_link='$CC -o conftest$ac_exeext $CFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5' ac_compiler_gnu=$ac_cv_c_compiler_gnu { $as_echo "$as_me:${as_lineno-$LINENO}: checking for ANSI C header files" >&5 $as_echo_n "checking for ANSI C header files... " >&6; } if ${ac_cv_header_stdc+:} false; then : $as_echo_n "(cached) " >&6 else cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ #include #include #include #include int main () { ; return 0; } _ACEOF if ac_fn_c_try_compile "$LINENO"; then : ac_cv_header_stdc=yes else ac_cv_header_stdc=no fi rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext if test $ac_cv_header_stdc = yes; then # SunOS 4.x string.h does not declare mem*, contrary to ANSI. cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ #include _ACEOF if (eval "$ac_cpp conftest.$ac_ext") 2>&5 | $EGREP "memchr" >/dev/null 2>&1; then : else ac_cv_header_stdc=no fi rm -f conftest* fi if test $ac_cv_header_stdc = yes; then # ISC 2.0.2 stdlib.h does not declare free, contrary to ANSI. cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ #include _ACEOF if (eval "$ac_cpp conftest.$ac_ext") 2>&5 | $EGREP "free" >/dev/null 2>&1; then : else ac_cv_header_stdc=no fi rm -f conftest* fi if test $ac_cv_header_stdc = yes; then # /bin/cc in Irix-4.0.5 gets non-ANSI ctype macros unless using -ansi. if test "$cross_compiling" = yes; then : : else cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ #include #include #if ((' ' & 0x0FF) == 0x020) # define ISLOWER(c) ('a' <= (c) && (c) <= 'z') # define TOUPPER(c) (ISLOWER(c) ? 'A' + ((c) - 'a') : (c)) #else # define ISLOWER(c) \ (('a' <= (c) && (c) <= 'i') \ || ('j' <= (c) && (c) <= 'r') \ || ('s' <= (c) && (c) <= 'z')) # define TOUPPER(c) (ISLOWER(c) ? ((c) | 0x40) : (c)) #endif #define XOR(e, f) (((e) && !(f)) || (!(e) && (f))) int main () { int i; for (i = 0; i < 256; i++) if (XOR (islower (i), ISLOWER (i)) || toupper (i) != TOUPPER (i)) return 2; return 0; } _ACEOF if ac_fn_c_try_run "$LINENO"; then : else ac_cv_header_stdc=no fi rm -f core *.core core.conftest.* gmon.out bb.out conftest$ac_exeext \ conftest.$ac_objext conftest.beam conftest.$ac_ext fi fi fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_cv_header_stdc" >&5 $as_echo "$ac_cv_header_stdc" >&6; } if test $ac_cv_header_stdc = yes; then $as_echo "#define STDC_HEADERS 1" >>confdefs.h fi # On IRIX 5.3, sys/types and inttypes.h are conflicting. for ac_header in sys/types.h sys/stat.h stdlib.h string.h memory.h strings.h \ inttypes.h stdint.h unistd.h do : as_ac_Header=`$as_echo "ac_cv_header_$ac_header" | $as_tr_sh` ac_fn_c_check_header_compile "$LINENO" "$ac_header" "$as_ac_Header" "$ac_includes_default " if eval test \"x\$"$as_ac_Header"\" = x"yes"; then : cat >>confdefs.h <<_ACEOF #define `$as_echo "HAVE_$ac_header" | $as_tr_cpp` 1 _ACEOF fi done for ac_header in dlfcn.h do : ac_fn_c_check_header_compile "$LINENO" "dlfcn.h" "ac_cv_header_dlfcn_h" "$ac_includes_default " if test "x$ac_cv_header_dlfcn_h" = xyes; then : cat >>confdefs.h <<_ACEOF #define HAVE_DLFCN_H 1 _ACEOF fi done func_stripname_cnf () { case $2 in .*) func_stripname_result=`$ECHO "$3" | $SED "s%^$1%%; s%\\\\$2\$%%"`;; *) func_stripname_result=`$ECHO "$3" | $SED "s%^$1%%; s%$2\$%%"`;; esac } # func_stripname_cnf # Set options enable_win32_dll=yes case $host in *-*-cygwin* | *-*-mingw* | *-*-pw32* | *-*-cegcc*) if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}as", so it can be a program name with args. set dummy ${ac_tool_prefix}as; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_AS+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$AS"; then ac_cv_prog_AS="$AS" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_AS="${ac_tool_prefix}as" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi AS=$ac_cv_prog_AS if test -n "$AS"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $AS" >&5 $as_echo "$AS" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_AS"; then ac_ct_AS=$AS # Extract the first word of "as", so it can be a program name with args. set dummy as; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_AS+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_AS"; then ac_cv_prog_ac_ct_AS="$ac_ct_AS" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_AS="as" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_AS=$ac_cv_prog_ac_ct_AS if test -n "$ac_ct_AS"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_AS" >&5 $as_echo "$ac_ct_AS" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_AS" = x; then AS="false" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac AS=$ac_ct_AS fi else AS="$ac_cv_prog_AS" fi if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}dlltool", so it can be a program name with args. set dummy ${ac_tool_prefix}dlltool; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_DLLTOOL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$DLLTOOL"; then ac_cv_prog_DLLTOOL="$DLLTOOL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_DLLTOOL="${ac_tool_prefix}dlltool" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi DLLTOOL=$ac_cv_prog_DLLTOOL if test -n "$DLLTOOL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $DLLTOOL" >&5 $as_echo "$DLLTOOL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_DLLTOOL"; then ac_ct_DLLTOOL=$DLLTOOL # Extract the first word of "dlltool", so it can be a program name with args. set dummy dlltool; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_DLLTOOL+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_DLLTOOL"; then ac_cv_prog_ac_ct_DLLTOOL="$ac_ct_DLLTOOL" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_DLLTOOL="dlltool" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_DLLTOOL=$ac_cv_prog_ac_ct_DLLTOOL if test -n "$ac_ct_DLLTOOL"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_DLLTOOL" >&5 $as_echo "$ac_ct_DLLTOOL" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_DLLTOOL" = x; then DLLTOOL="false" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac DLLTOOL=$ac_ct_DLLTOOL fi else DLLTOOL="$ac_cv_prog_DLLTOOL" fi if test -n "$ac_tool_prefix"; then # Extract the first word of "${ac_tool_prefix}objdump", so it can be a program name with args. set dummy ${ac_tool_prefix}objdump; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_OBJDUMP+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$OBJDUMP"; then ac_cv_prog_OBJDUMP="$OBJDUMP" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_OBJDUMP="${ac_tool_prefix}objdump" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi OBJDUMP=$ac_cv_prog_OBJDUMP if test -n "$OBJDUMP"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $OBJDUMP" >&5 $as_echo "$OBJDUMP" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi fi if test -z "$ac_cv_prog_OBJDUMP"; then ac_ct_OBJDUMP=$OBJDUMP # Extract the first word of "objdump", so it can be a program name with args. set dummy objdump; ac_word=$2 { $as_echo "$as_me:${as_lineno-$LINENO}: checking for $ac_word" >&5 $as_echo_n "checking for $ac_word... " >&6; } if ${ac_cv_prog_ac_ct_OBJDUMP+:} false; then : $as_echo_n "(cached) " >&6 else if test -n "$ac_ct_OBJDUMP"; then ac_cv_prog_ac_ct_OBJDUMP="$ac_ct_OBJDUMP" # Let the user override the test. else as_save_IFS=$IFS; IFS=$PATH_SEPARATOR for as_dir in $PATH do IFS=$as_save_IFS test -z "$as_dir" && as_dir=. for ac_exec_ext in '' $ac_executable_extensions; do if as_fn_executable_p "$as_dir/$ac_word$ac_exec_ext"; then ac_cv_prog_ac_ct_OBJDUMP="objdump" $as_echo "$as_me:${as_lineno-$LINENO}: found $as_dir/$ac_word$ac_exec_ext" >&5 break 2 fi done done IFS=$as_save_IFS fi fi ac_ct_OBJDUMP=$ac_cv_prog_ac_ct_OBJDUMP if test -n "$ac_ct_OBJDUMP"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $ac_ct_OBJDUMP" >&5 $as_echo "$ac_ct_OBJDUMP" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test "x$ac_ct_OBJDUMP" = x; then OBJDUMP="false" else case $cross_compiling:$ac_tool_warned in yes:) { $as_echo "$as_me:${as_lineno-$LINENO}: WARNING: using cross tools not prefixed with host triplet" >&5 $as_echo "$as_me: WARNING: using cross tools not prefixed with host triplet" >&2;} ac_tool_warned=yes ;; esac OBJDUMP=$ac_ct_OBJDUMP fi else OBJDUMP="$ac_cv_prog_OBJDUMP" fi ;; esac test -z "$AS" && AS=as test -z "$DLLTOOL" && DLLTOOL=dlltool test -z "$OBJDUMP" && OBJDUMP=objdump enable_dlopen=no # Check whether --enable-shared was given. if test "${enable_shared+set}" = set; then : enableval=$enable_shared; p=${PACKAGE-default} case $enableval in yes) enable_shared=yes ;; no) enable_shared=no ;; *) enable_shared=no # Look at the argument we got. We use all the common list separators. lt_save_ifs=$IFS; IFS=$IFS$PATH_SEPARATOR, for pkg in $enableval; do IFS=$lt_save_ifs if test "X$pkg" = "X$p"; then enable_shared=yes fi done IFS=$lt_save_ifs ;; esac else enable_shared=yes fi # Check whether --enable-static was given. if test "${enable_static+set}" = set; then : enableval=$enable_static; p=${PACKAGE-default} case $enableval in yes) enable_static=yes ;; no) enable_static=no ;; *) enable_static=no # Look at the argument we got. We use all the common list separators. lt_save_ifs=$IFS; IFS=$IFS$PATH_SEPARATOR, for pkg in $enableval; do IFS=$lt_save_ifs if test "X$pkg" = "X$p"; then enable_static=yes fi done IFS=$lt_save_ifs ;; esac else enable_static=yes fi # Check whether --with-pic was given. if test "${with_pic+set}" = set; then : withval=$with_pic; lt_p=${PACKAGE-default} case $withval in yes|no) pic_mode=$withval ;; *) pic_mode=default # Look at the argument we got. We use all the common list separators. lt_save_ifs=$IFS; IFS=$IFS$PATH_SEPARATOR, for lt_pkg in $withval; do IFS=$lt_save_ifs if test "X$lt_pkg" = "X$lt_p"; then pic_mode=yes fi done IFS=$lt_save_ifs ;; esac else pic_mode=default fi # Check whether --enable-fast-install was given. if test "${enable_fast_install+set}" = set; then : enableval=$enable_fast_install; p=${PACKAGE-default} case $enableval in yes) enable_fast_install=yes ;; no) enable_fast_install=no ;; *) enable_fast_install=no # Look at the argument we got. We use all the common list separators. lt_save_ifs=$IFS; IFS=$IFS$PATH_SEPARATOR, for pkg in $enableval; do IFS=$lt_save_ifs if test "X$pkg" = "X$p"; then enable_fast_install=yes fi done IFS=$lt_save_ifs ;; esac else enable_fast_install=yes fi shared_archive_member_spec= case $host,$enable_shared in power*-*-aix[5-9]*,yes) { $as_echo "$as_me:${as_lineno-$LINENO}: checking which variant of shared library versioning to provide" >&5 $as_echo_n "checking which variant of shared library versioning to provide... " >&6; } # Check whether --with-aix-soname was given. if test "${with_aix_soname+set}" = set; then : withval=$with_aix_soname; case $withval in aix|svr4|both) ;; *) as_fn_error $? "Unknown argument to --with-aix-soname" "$LINENO" 5 ;; esac lt_cv_with_aix_soname=$with_aix_soname else if ${lt_cv_with_aix_soname+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_with_aix_soname=aix fi with_aix_soname=$lt_cv_with_aix_soname fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $with_aix_soname" >&5 $as_echo "$with_aix_soname" >&6; } if test aix != "$with_aix_soname"; then # For the AIX way of multilib, we name the shared archive member # based on the bitwidth used, traditionally 'shr.o' or 'shr_64.o', # and 'shr.imp' or 'shr_64.imp', respectively, for the Import File. # Even when GNU compilers ignore OBJECT_MODE but need '-maix64' flag, # the AIX toolchain works better with OBJECT_MODE set (default 32). if test 64 = "${OBJECT_MODE-32}"; then shared_archive_member_spec=shr_64 else shared_archive_member_spec=shr fi fi ;; *) with_aix_soname=aix ;; esac # This can be used to rebuild libtool when needed LIBTOOL_DEPS=$ltmain # Always use our own libtool. LIBTOOL='$(SHELL) $(top_builddir)/libtool' test -z "$LN_S" && LN_S="ln -s" if test -n "${ZSH_VERSION+set}"; then setopt NO_GLOB_SUBST fi { $as_echo "$as_me:${as_lineno-$LINENO}: checking for objdir" >&5 $as_echo_n "checking for objdir... " >&6; } if ${lt_cv_objdir+:} false; then : $as_echo_n "(cached) " >&6 else rm -f .libs 2>/dev/null mkdir .libs 2>/dev/null if test -d .libs; then lt_cv_objdir=.libs else # MS-DOS does not allow filenames that begin with a dot. lt_cv_objdir=_libs fi rmdir .libs 2>/dev/null fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_objdir" >&5 $as_echo "$lt_cv_objdir" >&6; } objdir=$lt_cv_objdir cat >>confdefs.h <<_ACEOF #define LT_OBJDIR "$lt_cv_objdir/" _ACEOF case $host_os in aix3*) # AIX sometimes has problems with the GCC collect2 program. For some # reason, if we set the COLLECT_NAMES environment variable, the problems # vanish in a puff of smoke. if test set != "${COLLECT_NAMES+set}"; then COLLECT_NAMES= export COLLECT_NAMES fi ;; esac # Global variables: ofile=libtool can_build_shared=yes # All known linkers require a '.a' archive for static linking (except MSVC, # which needs '.lib'). libext=a with_gnu_ld=$lt_cv_prog_gnu_ld old_CC=$CC old_CFLAGS=$CFLAGS # Set sane defaults for various variables test -z "$CC" && CC=cc test -z "$LTCC" && LTCC=$CC test -z "$LTCFLAGS" && LTCFLAGS=$CFLAGS test -z "$LD" && LD=ld test -z "$ac_objext" && ac_objext=o func_cc_basename $compiler cc_basename=$func_cc_basename_result # Only perform the check for file, if the check method requires it test -z "$MAGIC_CMD" && MAGIC_CMD=file case $deplibs_check_method in file_magic*) if test "$file_magic_cmd" = '$MAGIC_CMD'; then { $as_echo "$as_me:${as_lineno-$LINENO}: checking for ${ac_tool_prefix}file" >&5 $as_echo_n "checking for ${ac_tool_prefix}file... " >&6; } if ${lt_cv_path_MAGIC_CMD+:} false; then : $as_echo_n "(cached) " >&6 else case $MAGIC_CMD in [\\/*] | ?:[\\/]*) lt_cv_path_MAGIC_CMD=$MAGIC_CMD # Let the user override the test with a path. ;; *) lt_save_MAGIC_CMD=$MAGIC_CMD lt_save_ifs=$IFS; IFS=$PATH_SEPARATOR ac_dummy="/usr/bin$PATH_SEPARATOR$PATH" for ac_dir in $ac_dummy; do IFS=$lt_save_ifs test -z "$ac_dir" && ac_dir=. if test -f "$ac_dir/${ac_tool_prefix}file"; then lt_cv_path_MAGIC_CMD=$ac_dir/"${ac_tool_prefix}file" if test -n "$file_magic_test_file"; then case $deplibs_check_method in "file_magic "*) file_magic_regex=`expr "$deplibs_check_method" : "file_magic \(.*\)"` MAGIC_CMD=$lt_cv_path_MAGIC_CMD if eval $file_magic_cmd \$file_magic_test_file 2> /dev/null | $EGREP "$file_magic_regex" > /dev/null; then : else cat <<_LT_EOF 1>&2 *** Warning: the command libtool uses to detect shared libraries, *** $file_magic_cmd, produces output that libtool cannot recognize. *** The result is that libtool may fail to recognize shared libraries *** as such. This will affect the creation of libtool libraries that *** depend on shared libraries, but programs linked with such libtool *** libraries will work regardless of this problem. Nevertheless, you *** may want to report the problem to your system manager and/or to *** bug-libtool@gnu.org _LT_EOF fi ;; esac fi break fi done IFS=$lt_save_ifs MAGIC_CMD=$lt_save_MAGIC_CMD ;; esac fi MAGIC_CMD=$lt_cv_path_MAGIC_CMD if test -n "$MAGIC_CMD"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $MAGIC_CMD" >&5 $as_echo "$MAGIC_CMD" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi if test -z "$lt_cv_path_MAGIC_CMD"; then if test -n "$ac_tool_prefix"; then { $as_echo "$as_me:${as_lineno-$LINENO}: checking for file" >&5 $as_echo_n "checking for file... " >&6; } if ${lt_cv_path_MAGIC_CMD+:} false; then : $as_echo_n "(cached) " >&6 else case $MAGIC_CMD in [\\/*] | ?:[\\/]*) lt_cv_path_MAGIC_CMD=$MAGIC_CMD # Let the user override the test with a path. ;; *) lt_save_MAGIC_CMD=$MAGIC_CMD lt_save_ifs=$IFS; IFS=$PATH_SEPARATOR ac_dummy="/usr/bin$PATH_SEPARATOR$PATH" for ac_dir in $ac_dummy; do IFS=$lt_save_ifs test -z "$ac_dir" && ac_dir=. if test -f "$ac_dir/file"; then lt_cv_path_MAGIC_CMD=$ac_dir/"file" if test -n "$file_magic_test_file"; then case $deplibs_check_method in "file_magic "*) file_magic_regex=`expr "$deplibs_check_method" : "file_magic \(.*\)"` MAGIC_CMD=$lt_cv_path_MAGIC_CMD if eval $file_magic_cmd \$file_magic_test_file 2> /dev/null | $EGREP "$file_magic_regex" > /dev/null; then : else cat <<_LT_EOF 1>&2 *** Warning: the command libtool uses to detect shared libraries, *** $file_magic_cmd, produces output that libtool cannot recognize. *** The result is that libtool may fail to recognize shared libraries *** as such. This will affect the creation of libtool libraries that *** depend on shared libraries, but programs linked with such libtool *** libraries will work regardless of this problem. Nevertheless, you *** may want to report the problem to your system manager and/or to *** bug-libtool@gnu.org _LT_EOF fi ;; esac fi break fi done IFS=$lt_save_ifs MAGIC_CMD=$lt_save_MAGIC_CMD ;; esac fi MAGIC_CMD=$lt_cv_path_MAGIC_CMD if test -n "$MAGIC_CMD"; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: $MAGIC_CMD" >&5 $as_echo "$MAGIC_CMD" >&6; } else { $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5 $as_echo "no" >&6; } fi else MAGIC_CMD=: fi fi fi ;; esac # Use C for the default configuration in the libtool script lt_save_CC=$CC ac_ext=c ac_cpp='$CPP $CPPFLAGS' ac_compile='$CC -c $CFLAGS $CPPFLAGS conftest.$ac_ext >&5' ac_link='$CC -o conftest$ac_exeext $CFLAGS $CPPFLAGS $LDFLAGS conftest.$ac_ext $LIBS >&5' ac_compiler_gnu=$ac_cv_c_compiler_gnu # Source file extension for C test sources. ac_ext=c # Object file extension for compiled C test sources. objext=o objext=$objext # Code to be used in simple compile tests lt_simple_compile_test_code="int some_variable = 0;" # Code to be used in simple link tests lt_simple_link_test_code='int main(){return(0);}' # If no C compiler was specified, use CC. LTCC=${LTCC-"$CC"} # If no C compiler flags were specified, use CFLAGS. LTCFLAGS=${LTCFLAGS-"$CFLAGS"} # Allow CC to be a program name with arguments. compiler=$CC # Save the default compiler, since it gets overwritten when the other # tags are being tested, and _LT_TAGVAR(compiler, []) is a NOP. compiler_DEFAULT=$CC # save warnings/boilerplate of simple test code ac_outfile=conftest.$ac_objext echo "$lt_simple_compile_test_code" >conftest.$ac_ext eval "$ac_compile" 2>&1 >/dev/null | $SED '/^$/d; /^ *+/d' >conftest.err _lt_compiler_boilerplate=`cat conftest.err` $RM conftest* ac_outfile=conftest.$ac_objext echo "$lt_simple_link_test_code" >conftest.$ac_ext eval "$ac_link" 2>&1 >/dev/null | $SED '/^$/d; /^ *+/d' >conftest.err _lt_linker_boilerplate=`cat conftest.err` $RM -r conftest* ## CAVEAT EMPTOR: ## There is no encapsulation within the following macros, do not change ## the running order or otherwise move them around unless you know exactly ## what you are doing... if test -n "$compiler"; then lt_prog_compiler_no_builtin_flag= if test yes = "$GCC"; then case $cc_basename in nvcc*) lt_prog_compiler_no_builtin_flag=' -Xcompiler -fno-builtin' ;; *) lt_prog_compiler_no_builtin_flag=' -fno-builtin' ;; esac { $as_echo "$as_me:${as_lineno-$LINENO}: checking if $compiler supports -fno-rtti -fno-exceptions" >&5 $as_echo_n "checking if $compiler supports -fno-rtti -fno-exceptions... " >&6; } if ${lt_cv_prog_compiler_rtti_exceptions+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_prog_compiler_rtti_exceptions=no ac_outfile=conftest.$ac_objext echo "$lt_simple_compile_test_code" > conftest.$ac_ext lt_compiler_flag="-fno-rtti -fno-exceptions" ## exclude from sc_useless_quotes_in_assignment # Insert the option either (1) after the last *FLAGS variable, or # (2) before a word containing "conftest.", or (3) at the end. # Note that $ac_compile itself does not contain backslashes and begins # with a dollar sign (not a hyphen), so the echo should work correctly. # The option is referenced via a variable to avoid confusing sed. lt_compile=`echo "$ac_compile" | $SED \ -e 's:.*FLAGS}\{0,1\} :&$lt_compiler_flag :; t' \ -e 's: [^ ]*conftest\.: $lt_compiler_flag&:; t' \ -e 's:$: $lt_compiler_flag:'` (eval echo "\"\$as_me:$LINENO: $lt_compile\"" >&5) (eval "$lt_compile" 2>conftest.err) ac_status=$? cat conftest.err >&5 echo "$as_me:$LINENO: \$? = $ac_status" >&5 if (exit $ac_status) && test -s "$ac_outfile"; then # The compiler can only warn and ignore the option if not recognized # So say no if there are warnings other than the usual output. $ECHO "$_lt_compiler_boilerplate" | $SED '/^$/d' >conftest.exp $SED '/^$/d; /^ *+/d' conftest.err >conftest.er2 if test ! -s conftest.er2 || diff conftest.exp conftest.er2 >/dev/null; then lt_cv_prog_compiler_rtti_exceptions=yes fi fi $RM conftest* fi { $as_echo "$as_me:${as_lineno-$LINENO}: result: $lt_cv_prog_compiler_rtti_exceptions" >&5 $as_echo "$lt_cv_prog_compiler_rtti_exceptions" >&6; } if test yes = "$lt_cv_prog_compiler_rtti_exceptions"; then lt_prog_compiler_no_builtin_flag="$lt_prog_compiler_no_builtin_flag -fno-rtti -fno-exceptions" else : fi fi lt_prog_compiler_wl= lt_prog_compiler_pic= lt_prog_compiler_static= if test yes = "$GCC"; then lt_prog_compiler_wl='-Wl,' lt_prog_compiler_static='-static' case $host_os in aix*) # All AIX code is PIC. if test ia64 = "$host_cpu"; then # AIX 5 now supports IA64 processor lt_prog_compiler_static='-Bstatic' fi lt_prog_compiler_pic='-fPIC' ;; amigaos*) case $host_cpu in powerpc) # see comment about AmigaOS4 .so support lt_prog_compiler_pic='-fPIC' ;; m68k) # FIXME: we need at least 68020 code to build shared libraries, but # adding the '-m68020' flag to GCC prevents building anything better, # like '-m68040'. lt_prog_compiler_pic='-m68020 -resident32 -malways-restore-a4' ;; esac ;; beos* | irix5* | irix6* | nonstopux* | osf3* | osf4* | osf5*) # PIC is the default for these OSes. ;; mingw* | cygwin* | pw32* | os2* | cegcc*) # This hack is so that the source file can tell whether it is being # built for inclusion in a dll (and should export symbols for example). # Although the cygwin gcc ignores -fPIC, still need this for old-style # (--disable-auto-import) libraries lt_prog_compiler_pic='-DDLL_EXPORT' case $host_os in os2*) lt_prog_compiler_static='$wl-static' ;; esac ;; darwin* | rhapsody*) # PIC is the default on this platform # Common symbols not allowed in MH_DYLIB files lt_prog_compiler_pic='-fno-common' ;; haiku*) # PIC is the default for Haiku. # The "-static" flag exists, but is broken. lt_prog_compiler_static= ;; hpux*) # PIC is the default for 64-bit PA HP-UX, but not for 32-bit # PA HP-UX. 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If you *** really care for shared libraries, you may want to install binutils *** 2.20 or above, or modify your PATH so that a non-GNU linker is found. *** You will then need to restart the configuration process. _LT_EOF fi ;; amigaos*) case $host_cpu in powerpc) # see comment about AmigaOS4 .so support archive_cmds='$CC -shared $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' archive_expsym_cmds='' ;; m68k) archive_cmds='$RM $output_objdir/a2ixlibrary.data~$ECHO "#define NAME $libname" > $output_objdir/a2ixlibrary.data~$ECHO "#define LIBRARY_ID 1" >> $output_objdir/a2ixlibrary.data~$ECHO "#define VERSION $major" >> $output_objdir/a2ixlibrary.data~$ECHO "#define REVISION $revision" >> $output_objdir/a2ixlibrary.data~$AR $AR_FLAGS $lib $libobjs~$RANLIB $lib~(cd $output_objdir && a2ixlibrary -32)' hardcode_libdir_flag_spec='-L$libdir' hardcode_minus_L=yes ;; esac ;; beos*) if $LD --help 2>&1 | $GREP ': supported targets:.* elf' > /dev/null; then allow_undefined_flag=unsupported # Joseph Beckenbach says some releases of gcc # support --undefined. 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FIXME archive_cmds='$CC -nostart $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' else ld_shlibs=no fi ;; cygwin* | mingw* | pw32* | cegcc*) # _LT_TAGVAR(hardcode_libdir_flag_spec, ) is actually meaningless, # as there is no search path for DLLs. hardcode_libdir_flag_spec='-L$libdir' export_dynamic_flag_spec='$wl--export-all-symbols' allow_undefined_flag=unsupported always_export_symbols=no enable_shared_with_static_runtimes=yes export_symbols_cmds='$NM $libobjs $convenience | $global_symbol_pipe | $SED -e '\''/^[BCDGRS][ ]/s/.*[ ]\([^ ]*\)/\1 DATA/;s/^.*[ ]__nm__\([^ ]*\)[ ][^ ]*/\1 DATA/;/^I[ ]/d;/^[AITW][ ]/s/.* //'\'' | sort | uniq > $export_symbols' exclude_expsyms='[_]+GLOBAL_OFFSET_TABLE_|[_]+GLOBAL__[FID]_.*|[_]+head_[A-Za-z0-9_]+_dll|[A-Za-z0-9_]+_dll_iname' if $LD --help 2>&1 | $GREP 'auto-import' > /dev/null; then archive_cmds='$CC -shared $libobjs $deplibs $compiler_flags -o $output_objdir/$soname $wl--enable-auto-image-base -Xlinker --out-implib -Xlinker $lib' # If the export-symbols file already is a .def file, use it as # is; otherwise, prepend EXPORTS... archive_expsym_cmds='if test DEF = "`$SED -n -e '\''s/^[ ]*//'\'' -e '\''/^\(;.*\)*$/d'\'' -e '\''s/^\(EXPORTS\|LIBRARY\)\([ ].*\)*$/DEF/p'\'' -e q $export_symbols`" ; then cp $export_symbols $output_objdir/$soname.def; else echo EXPORTS > $output_objdir/$soname.def; cat $export_symbols >> $output_objdir/$soname.def; fi~ $CC -shared $output_objdir/$soname.def $libobjs $deplibs $compiler_flags -o $output_objdir/$soname $wl--enable-auto-image-base -Xlinker --out-implib -Xlinker $lib' else ld_shlibs=no fi ;; haiku*) archive_cmds='$CC -shared $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' link_all_deplibs=yes ;; os2*) hardcode_libdir_flag_spec='-L$libdir' hardcode_minus_L=yes allow_undefined_flag=unsupported shrext_cmds=.dll archive_cmds='$ECHO "LIBRARY ${soname%$shared_ext} INITINSTANCE TERMINSTANCE" > $output_objdir/$libname.def~ $ECHO "DESCRIPTION \"$libname\"" >> $output_objdir/$libname.def~ $ECHO "DATA MULTIPLE NONSHARED" >> $output_objdir/$libname.def~ $ECHO EXPORTS >> $output_objdir/$libname.def~ emxexp $libobjs | $SED /"_DLL_InitTerm"/d >> $output_objdir/$libname.def~ $CC -Zdll -Zcrtdll -o $output_objdir/$soname $libobjs $deplibs $compiler_flags $output_objdir/$libname.def~ emximp -o $lib $output_objdir/$libname.def' archive_expsym_cmds='$ECHO "LIBRARY ${soname%$shared_ext} INITINSTANCE TERMINSTANCE" > $output_objdir/$libname.def~ $ECHO "DESCRIPTION \"$libname\"" >> $output_objdir/$libname.def~ $ECHO "DATA MULTIPLE NONSHARED" >> $output_objdir/$libname.def~ $ECHO EXPORTS >> $output_objdir/$libname.def~ prefix_cmds="$SED"~ if test EXPORTS = "`$SED 1q $export_symbols`"; then prefix_cmds="$prefix_cmds -e 1d"; fi~ prefix_cmds="$prefix_cmds -e \"s/^\(.*\)$/_\1/g\""~ cat $export_symbols | $prefix_cmds >> $output_objdir/$libname.def~ $CC -Zdll -Zcrtdll -o $output_objdir/$soname $libobjs $deplibs $compiler_flags $output_objdir/$libname.def~ emximp -o $lib $output_objdir/$libname.def' old_archive_From_new_cmds='emximp -o $output_objdir/${libname}_dll.a $output_objdir/$libname.def' enable_shared_with_static_runtimes=yes ;; interix[3-9]*) hardcode_direct=no hardcode_shlibpath_var=no hardcode_libdir_flag_spec='$wl-rpath,$libdir' export_dynamic_flag_spec='$wl-E' # Hack: On Interix 3.x, we cannot compile PIC because of a broken gcc. # Instead, shared libraries are loaded at an image base (0x10000000 by # default) and relocated if they conflict, which is a slow very memory # consuming and fragmenting process. To avoid this, we pick a random, # 256 KiB-aligned image base between 0x50000000 and 0x6FFC0000 at link # time. Moving up from 0x10000000 also allows more sbrk(2) space. archive_cmds='$CC -shared $pic_flag $libobjs $deplibs $compiler_flags $wl-h,$soname $wl--image-base,`expr ${RANDOM-$$} % 4096 / 2 \* 262144 + 1342177280` -o $lib' archive_expsym_cmds='sed "s|^|_|" $export_symbols >$output_objdir/$soname.expsym~$CC -shared $pic_flag $libobjs $deplibs $compiler_flags $wl-h,$soname $wl--retain-symbols-file,$output_objdir/$soname.expsym $wl--image-base,`expr ${RANDOM-$$} % 4096 / 2 \* 262144 + 1342177280` -o $lib' ;; gnu* | linux* | tpf* | k*bsd*-gnu | kopensolaris*-gnu) tmp_diet=no if test linux-dietlibc = "$host_os"; then case $cc_basename in diet\ *) tmp_diet=yes;; # linux-dietlibc with static linking (!diet-dyn) esac fi if $LD --help 2>&1 | $EGREP ': supported targets:.* elf' > /dev/null \ && test no = "$tmp_diet" then tmp_addflag=' $pic_flag' tmp_sharedflag='-shared' case $cc_basename,$host_cpu in pgcc*) # Portland Group C compiler whole_archive_flag_spec='$wl--whole-archive`for conv in $convenience\"\"; do test -n \"$conv\" && new_convenience=\"$new_convenience,$conv\"; done; func_echo_all \"$new_convenience\"` $wl--no-whole-archive' tmp_addflag=' $pic_flag' ;; pgf77* | pgf90* | pgf95* | pgfortran*) # Portland Group f77 and f90 compilers whole_archive_flag_spec='$wl--whole-archive`for conv in $convenience\"\"; do test -n \"$conv\" && new_convenience=\"$new_convenience,$conv\"; done; func_echo_all \"$new_convenience\"` $wl--no-whole-archive' tmp_addflag=' $pic_flag -Mnomain' ;; ecc*,ia64* | icc*,ia64*) # Intel C compiler on ia64 tmp_addflag=' -i_dynamic' ;; efc*,ia64* | ifort*,ia64*) # Intel Fortran compiler on ia64 tmp_addflag=' -i_dynamic -nofor_main' ;; ifc* | ifort*) # Intel Fortran compiler tmp_addflag=' -nofor_main' ;; lf95*) # Lahey Fortran 8.1 whole_archive_flag_spec= tmp_sharedflag='--shared' ;; nagfor*) # NAGFOR 5.3 tmp_sharedflag='-Wl,-shared' ;; xl[cC]* | bgxl[cC]* | mpixl[cC]*) # IBM XL C 8.0 on PPC (deal with xlf below) tmp_sharedflag='-qmkshrobj' tmp_addflag= ;; nvcc*) # Cuda Compiler Driver 2.2 whole_archive_flag_spec='$wl--whole-archive`for conv in $convenience\"\"; do test -n \"$conv\" && new_convenience=\"$new_convenience,$conv\"; done; func_echo_all \"$new_convenience\"` $wl--no-whole-archive' compiler_needs_object=yes ;; esac case `$CC -V 2>&1 | sed 5q` in *Sun\ C*) # Sun C 5.9 whole_archive_flag_spec='$wl--whole-archive`new_convenience=; for conv in $convenience\"\"; do test -z \"$conv\" || new_convenience=\"$new_convenience,$conv\"; done; func_echo_all \"$new_convenience\"` $wl--no-whole-archive' compiler_needs_object=yes tmp_sharedflag='-G' ;; *Sun\ F*) # Sun Fortran 8.3 tmp_sharedflag='-G' ;; esac archive_cmds='$CC '"$tmp_sharedflag""$tmp_addflag"' $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' if test yes = "$supports_anon_versioning"; then archive_expsym_cmds='echo "{ global:" > $output_objdir/$libname.ver~ cat $export_symbols | sed -e "s/\(.*\)/\1;/" >> $output_objdir/$libname.ver~ echo "local: *; };" >> $output_objdir/$libname.ver~ $CC '"$tmp_sharedflag""$tmp_addflag"' $libobjs $deplibs $compiler_flags $wl-soname $wl$soname $wl-version-script $wl$output_objdir/$libname.ver -o $lib' fi case $cc_basename in tcc*) export_dynamic_flag_spec='-rdynamic' ;; xlf* | bgf* | bgxlf* | mpixlf*) # IBM XL Fortran 10.1 on PPC cannot create shared libs itself whole_archive_flag_spec='--whole-archive$convenience --no-whole-archive' hardcode_libdir_flag_spec='$wl-rpath $wl$libdir' archive_cmds='$LD -shared $libobjs $deplibs $linker_flags -soname $soname -o $lib' if test yes = "$supports_anon_versioning"; then archive_expsym_cmds='echo "{ global:" > $output_objdir/$libname.ver~ cat $export_symbols | sed -e "s/\(.*\)/\1;/" >> $output_objdir/$libname.ver~ echo "local: *; };" >> $output_objdir/$libname.ver~ $LD -shared $libobjs $deplibs $linker_flags -soname $soname -version-script $output_objdir/$libname.ver -o $lib' fi ;; esac else ld_shlibs=no fi ;; netbsd*) if echo __ELF__ | $CC -E - | $GREP __ELF__ >/dev/null; then archive_cmds='$LD -Bshareable $libobjs $deplibs $linker_flags -o $lib' wlarc= else archive_cmds='$CC -shared $pic_flag $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' archive_expsym_cmds='$CC -shared $pic_flag $libobjs $deplibs $compiler_flags $wl-soname $wl$soname $wl-retain-symbols-file $wl$export_symbols -o $lib' fi ;; solaris*) if $LD -v 2>&1 | $GREP 'BFD 2\.8' > /dev/null; then ld_shlibs=no cat <<_LT_EOF 1>&2 *** Warning: The releases 2.8.* of the GNU linker cannot reliably *** create shared libraries on Solaris systems. Therefore, libtool *** is disabling shared libraries support. We urge you to upgrade GNU *** binutils to release 2.9.1 or newer. Another option is to modify *** your PATH or compiler configuration so that the native linker is *** used, and then restart. _LT_EOF elif $LD --help 2>&1 | $GREP ': supported targets:.* elf' > /dev/null; then archive_cmds='$CC -shared $pic_flag $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' archive_expsym_cmds='$CC -shared $pic_flag $libobjs $deplibs $compiler_flags $wl-soname $wl$soname $wl-retain-symbols-file $wl$export_symbols -o $lib' else ld_shlibs=no fi ;; sysv5* | sco3.2v5* | sco5v6* | unixware* | OpenUNIX*) case `$LD -v 2>&1` in *\ [01].* | *\ 2.[0-9].* | *\ 2.1[0-5].*) ld_shlibs=no cat <<_LT_EOF 1>&2 *** Warning: Releases of the GNU linker prior to 2.16.91.0.3 cannot *** reliably create shared libraries on SCO systems. Therefore, libtool *** is disabling shared libraries support. We urge you to upgrade GNU *** binutils to release 2.16.91.0.3 or newer. Another option is to modify *** your PATH or compiler configuration so that the native linker is *** used, and then restart. _LT_EOF ;; *) # For security reasons, it is highly recommended that you always # use absolute paths for naming shared libraries, and exclude the # DT_RUNPATH tag from executables and libraries. But doing so # requires that you compile everything twice, which is a pain. if $LD --help 2>&1 | $GREP ': supported targets:.* elf' > /dev/null; then hardcode_libdir_flag_spec='$wl-rpath $wl$libdir' archive_cmds='$CC -shared $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' archive_expsym_cmds='$CC -shared $libobjs $deplibs $compiler_flags $wl-soname $wl$soname $wl-retain-symbols-file $wl$export_symbols -o $lib' else ld_shlibs=no fi ;; esac ;; sunos4*) archive_cmds='$LD -assert pure-text -Bshareable -o $lib $libobjs $deplibs $linker_flags' wlarc= hardcode_direct=yes hardcode_shlibpath_var=no ;; *) if $LD --help 2>&1 | $GREP ': supported targets:.* elf' > /dev/null; then archive_cmds='$CC -shared $pic_flag $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' archive_expsym_cmds='$CC -shared $pic_flag $libobjs $deplibs $compiler_flags $wl-soname $wl$soname $wl-retain-symbols-file $wl$export_symbols -o $lib' else ld_shlibs=no fi ;; esac if test no = "$ld_shlibs"; then runpath_var= hardcode_libdir_flag_spec= export_dynamic_flag_spec= whole_archive_flag_spec= fi else # PORTME fill in a description of your system's linker (not GNU ld) case $host_os in aix3*) allow_undefined_flag=unsupported always_export_symbols=yes archive_expsym_cmds='$LD -o $output_objdir/$soname $libobjs $deplibs $linker_flags -bE:$export_symbols -T512 -H512 -bM:SRE~$AR $AR_FLAGS $lib $output_objdir/$soname' # Note: this linker hardcodes the directories in LIBPATH if there # are no directories specified by -L. hardcode_minus_L=yes if test yes = "$GCC" && test -z "$lt_prog_compiler_static"; then # Neither direct hardcoding nor static linking is supported with a # broken collect2. hardcode_direct=unsupported fi ;; aix[4-9]*) if test ia64 = "$host_cpu"; then # On IA64, the linker does run time linking by default, so we don't # have to do anything special. aix_use_runtimelinking=no exp_sym_flag='-Bexport' no_entry_flag= else # If we're using GNU nm, then we don't want the "-C" option. # -C means demangle to GNU nm, but means don't demangle to AIX nm. # Without the "-l" option, or with the "-B" option, AIX nm treats # weak defined symbols like other global defined symbols, whereas # GNU nm marks them as "W". # While the 'weak' keyword is ignored in the Export File, we need # it in the Import File for the 'aix-soname' feature, so we have # to replace the "-B" option with "-P" for AIX nm. if $NM -V 2>&1 | $GREP 'GNU' > /dev/null; then export_symbols_cmds='$NM -Bpg $libobjs $convenience | awk '\''{ if (((\$ 2 == "T") || (\$ 2 == "D") || (\$ 2 == "B") || (\$ 2 == "W")) && (substr(\$ 3,1,1) != ".")) { if (\$ 2 == "W") { print \$ 3 " weak" } else { print \$ 3 } } }'\'' | sort -u > $export_symbols' else export_symbols_cmds='`func_echo_all $NM | $SED -e '\''s/B\([^B]*\)$/P\1/'\''` -PCpgl $libobjs $convenience | awk '\''{ if (((\$ 2 == "T") || (\$ 2 == "D") || (\$ 2 == "B") || (\$ 2 == "W") || (\$ 2 == "V") || (\$ 2 == "Z")) && (substr(\$ 1,1,1) != ".")) { if ((\$ 2 == "W") || (\$ 2 == "V") || (\$ 2 == "Z")) { print \$ 1 " weak" } else { print \$ 1 } } }'\'' | sort -u > $export_symbols' fi aix_use_runtimelinking=no # Test if we are trying to use run time linking or normal # AIX style linking. If -brtl is somewhere in LDFLAGS, we # have runtime linking enabled, and use it for executables. # For shared libraries, we enable/disable runtime linking # depending on the kind of the shared library created - # when "with_aix_soname,aix_use_runtimelinking" is: # "aix,no" lib.a(lib.so.V) shared, rtl:no, for executables # "aix,yes" lib.so shared, rtl:yes, for executables # lib.a static archive # "both,no" lib.so.V(shr.o) shared, rtl:yes # lib.a(lib.so.V) shared, rtl:no, for executables # "both,yes" lib.so.V(shr.o) shared, rtl:yes, for executables # lib.a(lib.so.V) shared, rtl:no # "svr4,*" lib.so.V(shr.o) shared, rtl:yes, for executables # lib.a static archive case $host_os in aix4.[23]|aix4.[23].*|aix[5-9]*) for ld_flag in $LDFLAGS; do if (test x-brtl = "x$ld_flag" || test x-Wl,-brtl = "x$ld_flag"); then aix_use_runtimelinking=yes break fi done if test svr4,no = "$with_aix_soname,$aix_use_runtimelinking"; then # With aix-soname=svr4, we create the lib.so.V shared archives only, # so we don't have lib.a shared libs to link our executables. # We have to force runtime linking in this case. aix_use_runtimelinking=yes LDFLAGS="$LDFLAGS -Wl,-brtl" fi ;; esac exp_sym_flag='-bexport' no_entry_flag='-bnoentry' fi # When large executables or shared objects are built, AIX ld can # have problems creating the table of contents. If linking a library # or program results in "error TOC overflow" add -mminimal-toc to # CXXFLAGS/CFLAGS for g++/gcc. In the cases where that is not # enough to fix the problem, add -Wl,-bbigtoc to LDFLAGS. archive_cmds='' hardcode_direct=yes hardcode_direct_absolute=yes hardcode_libdir_separator=':' link_all_deplibs=yes file_list_spec='$wl-f,' case $with_aix_soname,$aix_use_runtimelinking in aix,*) ;; # traditional, no import file svr4,* | *,yes) # use import file # The Import File defines what to hardcode. hardcode_direct=no hardcode_direct_absolute=no ;; esac if test yes = "$GCC"; then case $host_os in aix4.[012]|aix4.[012].*) # We only want to do this on AIX 4.2 and lower, the check # below for broken collect2 doesn't work under 4.3+ collect2name=`$CC -print-prog-name=collect2` if test -f "$collect2name" && strings "$collect2name" | $GREP resolve_lib_name >/dev/null then # We have reworked collect2 : else # We have old collect2 hardcode_direct=unsupported # It fails to find uninstalled libraries when the uninstalled # path is not listed in the libpath. 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then : lt_aix_libpath_sed=' /Import File Strings/,/^$/ { /^0/ { s/^0 *\([^ ]*\) *$/\1/ p } }' lt_cv_aix_libpath_=`dump -H conftest$ac_exeext 2>/dev/null | $SED -n -e "$lt_aix_libpath_sed"` # Check for a 64-bit object if we didn't find anything. if test -z "$lt_cv_aix_libpath_"; then lt_cv_aix_libpath_=`dump -HX64 conftest$ac_exeext 2>/dev/null | $SED -n -e "$lt_aix_libpath_sed"` fi fi rm -f core conftest.err conftest.$ac_objext \ conftest$ac_exeext conftest.$ac_ext if test -z "$lt_cv_aix_libpath_"; then lt_cv_aix_libpath_=/usr/lib:/lib fi fi aix_libpath=$lt_cv_aix_libpath_ fi hardcode_libdir_flag_spec='$wl-blibpath:$libdir:'"$aix_libpath" archive_expsym_cmds='$CC -o $output_objdir/$soname $libobjs $deplibs $wl'$no_entry_flag' $compiler_flags `if test -n "$allow_undefined_flag"; then func_echo_all "$wl$allow_undefined_flag"; else :; fi` $wl'$exp_sym_flag:\$export_symbols' '$shared_flag else if test ia64 = "$host_cpu"; then hardcode_libdir_flag_spec='$wl-R $libdir:/usr/lib:/lib' allow_undefined_flag="-z nodefs" archive_expsym_cmds="\$CC $shared_flag"' -o $output_objdir/$soname $libobjs $deplibs '"\$wl$no_entry_flag"' $compiler_flags $wl$allow_undefined_flag '"\$wl$exp_sym_flag:\$export_symbols" else # Determine the default libpath from the value encoded in an # empty executable. if test set = "${lt_cv_aix_libpath+set}"; then aix_libpath=$lt_cv_aix_libpath else if ${lt_cv_aix_libpath_+:} false; then : $as_echo_n "(cached) " >&6 else cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ int main () { ; return 0; } _ACEOF if ac_fn_c_try_link "$LINENO"; then : lt_aix_libpath_sed=' /Import File Strings/,/^$/ { /^0/ { s/^0 *\([^ ]*\) *$/\1/ p } }' lt_cv_aix_libpath_=`dump -H conftest$ac_exeext 2>/dev/null | $SED -n -e "$lt_aix_libpath_sed"` # Check for a 64-bit object if we didn't find anything. if test -z "$lt_cv_aix_libpath_"; then lt_cv_aix_libpath_=`dump -HX64 conftest$ac_exeext 2>/dev/null | $SED -n -e "$lt_aix_libpath_sed"` fi fi rm -f core conftest.err conftest.$ac_objext \ conftest$ac_exeext conftest.$ac_ext if test -z "$lt_cv_aix_libpath_"; then lt_cv_aix_libpath_=/usr/lib:/lib fi fi aix_libpath=$lt_cv_aix_libpath_ fi hardcode_libdir_flag_spec='$wl-blibpath:$libdir:'"$aix_libpath" # Warning - without using the other run time loading flags, # -berok will link without error, but may produce a broken library. no_undefined_flag=' $wl-bernotok' allow_undefined_flag=' $wl-berok' if test yes = "$with_gnu_ld"; then # We only use this code for GNU lds that support --whole-archive. whole_archive_flag_spec='$wl--whole-archive$convenience $wl--no-whole-archive' else # Exported symbols can be pulled into shared objects from archives whole_archive_flag_spec='$convenience' fi archive_cmds_need_lc=yes archive_expsym_cmds='$RM -r $output_objdir/$realname.d~$MKDIR $output_objdir/$realname.d' # -brtl affects multiple linker settings, -berok does not and is overridden later compiler_flags_filtered='`func_echo_all "$compiler_flags " | $SED -e "s%-brtl\\([, ]\\)%-berok\\1%g"`' if test svr4 != "$with_aix_soname"; then # This is similar to how AIX traditionally builds its shared libraries. archive_expsym_cmds="$archive_expsym_cmds"'~$CC '$shared_flag_aix' -o $output_objdir/$realname.d/$soname $libobjs $deplibs $wl-bnoentry '$compiler_flags_filtered'$wl-bE:$export_symbols$allow_undefined_flag~$AR $AR_FLAGS $output_objdir/$libname$release.a $output_objdir/$realname.d/$soname' fi if test aix != "$with_aix_soname"; then archive_expsym_cmds="$archive_expsym_cmds"'~$CC '$shared_flag_svr4' -o $output_objdir/$realname.d/$shared_archive_member_spec.o $libobjs $deplibs $wl-bnoentry '$compiler_flags_filtered'$wl-bE:$export_symbols$allow_undefined_flag~$STRIP -e $output_objdir/$realname.d/$shared_archive_member_spec.o~( func_echo_all "#! $soname($shared_archive_member_spec.o)"; if test shr_64 = "$shared_archive_member_spec"; then func_echo_all "# 64"; else func_echo_all "# 32"; fi; cat $export_symbols ) > $output_objdir/$realname.d/$shared_archive_member_spec.imp~$AR $AR_FLAGS $output_objdir/$soname $output_objdir/$realname.d/$shared_archive_member_spec.o $output_objdir/$realname.d/$shared_archive_member_spec.imp' else # used by -dlpreopen to get the symbols archive_expsym_cmds="$archive_expsym_cmds"'~$MV $output_objdir/$realname.d/$soname $output_objdir' fi archive_expsym_cmds="$archive_expsym_cmds"'~$RM -r $output_objdir/$realname.d' fi fi ;; amigaos*) case $host_cpu in powerpc) # see comment about AmigaOS4 .so support archive_cmds='$CC -shared $libobjs $deplibs $compiler_flags $wl-soname $wl$soname -o $lib' archive_expsym_cmds='' ;; m68k) archive_cmds='$RM $output_objdir/a2ixlibrary.data~$ECHO "#define NAME $libname" > $output_objdir/a2ixlibrary.data~$ECHO "#define LIBRARY_ID 1" >> $output_objdir/a2ixlibrary.data~$ECHO "#define VERSION $major" >> $output_objdir/a2ixlibrary.data~$ECHO "#define REVISION $revision" >> $output_objdir/a2ixlibrary.data~$AR $AR_FLAGS $lib $libobjs~$RANLIB $lib~(cd $output_objdir && a2ixlibrary -32)' hardcode_libdir_flag_spec='-L$libdir' hardcode_minus_L=yes ;; esac ;; bsdi[45]*) export_dynamic_flag_spec=-rdynamic ;; cygwin* | mingw* | pw32* | cegcc*) # When not using gcc, we currently assume that we are using # Microsoft Visual C++. # hardcode_libdir_flag_spec is actually meaningless, as there is # no search path for DLLs. case $cc_basename in cl*) # Native MSVC hardcode_libdir_flag_spec=' ' allow_undefined_flag=unsupported always_export_symbols=yes file_list_spec='@' # Tell ltmain to make .lib files, not .a files. libext=lib # Tell ltmain to make .dll files, not .so files. shrext_cmds=.dll # FIXME: Setting linknames here is a bad hack. archive_cmds='$CC -o $output_objdir/$soname $libobjs $compiler_flags $deplibs -Wl,-DLL,-IMPLIB:"$tool_output_objdir$libname.dll.lib"~linknames=' archive_expsym_cmds='if test DEF = "`$SED -n -e '\''s/^[ ]*//'\'' -e '\''/^\(;.*\)*$/d'\'' -e '\''s/^\(EXPORTS\|LIBRARY\)\([ ].*\)*$/DEF/p'\'' -e q $export_symbols`" ; then cp "$export_symbols" "$output_objdir/$soname.def"; echo "$tool_output_objdir$soname.def" > "$output_objdir/$soname.exp"; else $SED -e '\''s/^/-link -EXPORT:/'\'' < $export_symbols > $output_objdir/$soname.exp; fi~ $CC -o $tool_output_objdir$soname $libobjs $compiler_flags $deplibs "@$tool_output_objdir$soname.exp" -Wl,-DLL,-IMPLIB:"$tool_output_objdir$libname.dll.lib"~ linknames=' # The linker will not automatically build a static lib if we build a DLL. # _LT_TAGVAR(old_archive_from_new_cmds, )='true' enable_shared_with_static_runtimes=yes exclude_expsyms='_NULL_IMPORT_DESCRIPTOR|_IMPORT_DESCRIPTOR_.*' export_symbols_cmds='$NM $libobjs $convenience | $global_symbol_pipe | $SED -e '\''/^[BCDGRS][ ]/s/.*[ ]\([^ ]*\)/\1,DATA/'\'' | $SED -e '\''/^[AITW][ ]/s/.*[ ]//'\'' | sort | uniq > $export_symbols' # Don't use ranlib old_postinstall_cmds='chmod 644 $oldlib' postlink_cmds='lt_outputfile="@OUTPUT@"~ lt_tool_outputfile="@TOOL_OUTPUT@"~ case $lt_outputfile in *.exe|*.EXE) ;; *) lt_outputfile=$lt_outputfile.exe lt_tool_outputfile=$lt_tool_outputfile.exe ;; esac~ if test : != "$MANIFEST_TOOL" && test -f "$lt_outputfile.manifest"; then $MANIFEST_TOOL -manifest "$lt_tool_outputfile.manifest" -outputresource:"$lt_tool_outputfile" || exit 1; $RM "$lt_outputfile.manifest"; fi' ;; *) # Assume MSVC wrapper hardcode_libdir_flag_spec=' ' allow_undefined_flag=unsupported # Tell ltmain to make .lib files, not .a files. libext=lib # Tell ltmain to make .dll files, not .so files. shrext_cmds=.dll # FIXME: Setting linknames here is a bad hack. archive_cmds='$CC -o $lib $libobjs $compiler_flags `func_echo_all "$deplibs" | $SED '\''s/ -lc$//'\''` -link -dll~linknames=' # The linker will automatically build a .lib file if we build a DLL. old_archive_from_new_cmds='true' # FIXME: Should let the user specify the lib program. old_archive_cmds='lib -OUT:$oldlib$oldobjs$old_deplibs' enable_shared_with_static_runtimes=yes ;; esac ;; darwin* | rhapsody*) archive_cmds_need_lc=no hardcode_direct=no hardcode_automatic=yes hardcode_shlibpath_var=unsupported if test yes = "$lt_cv_ld_force_load"; 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dgux*) archive_cmds='$LD -G -h $soname -o $lib $libobjs $deplibs $linker_flags' hardcode_libdir_flag_spec='-L$libdir' hardcode_shlibpath_var=no ;; # FreeBSD 2.2.[012] allows us to include c++rt0.o to get C++ constructor # support. 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linux*) case $cc_basename in tcc*) # Fabrice Bellard et al's Tiny C Compiler ld_shlibs=yes archive_cmds='$CC -shared $pic_flag -o $lib $libobjs $deplibs $compiler_flags' ;; esac ;; netbsd*) if echo __ELF__ | $CC -E - | $GREP __ELF__ >/dev/null; then archive_cmds='$LD -Bshareable -o $lib $libobjs $deplibs $linker_flags' # a.out else archive_cmds='$LD -shared -o $lib $libobjs $deplibs $linker_flags' # ELF fi hardcode_libdir_flag_spec='-R$libdir' hardcode_direct=yes hardcode_shlibpath_var=no ;; newsos6) archive_cmds='$LD -G -h $soname -o $lib $libobjs $deplibs $linker_flags' hardcode_direct=yes hardcode_libdir_flag_spec='$wl-rpath $wl$libdir' hardcode_libdir_separator=: hardcode_shlibpath_var=no ;; *nto* | *qnx*) ;; openbsd* | bitrig*) if test -f /usr/libexec/ld.so; then hardcode_direct=yes hardcode_shlibpath_var=no hardcode_direct_absolute=yes if test -z "`echo __ELF__ | $CC -E - | $GREP __ELF__`"; then archive_cmds='$CC -shared $pic_flag -o $lib $libobjs $deplibs $compiler_flags' archive_expsym_cmds='$CC -shared $pic_flag -o $lib $libobjs $deplibs $compiler_flags $wl-retain-symbols-file,$export_symbols' hardcode_libdir_flag_spec='$wl-rpath,$libdir' export_dynamic_flag_spec='$wl-E' else archive_cmds='$CC -shared $pic_flag -o $lib $libobjs $deplibs $compiler_flags' hardcode_libdir_flag_spec='$wl-rpath,$libdir' fi else ld_shlibs=no fi ;; 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When/if they implement a new # versioning mechanism, adjust this. if test -x /usr/bin/objformat; then objformat=`/usr/bin/objformat` else case $host_os in freebsd[23].*) objformat=aout ;; *) objformat=elf ;; esac fi version_type=freebsd-$objformat case $version_type in freebsd-elf*) library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' need_version=no need_lib_prefix=no ;; freebsd-*) library_names_spec='$libname$release$shared_ext$versuffix $libname$shared_ext$versuffix' need_version=yes ;; esac shlibpath_var=LD_LIBRARY_PATH case $host_os in freebsd2.*) shlibpath_overrides_runpath=yes ;; freebsd3.[01]* | freebsdelf3.[01]*) shlibpath_overrides_runpath=yes hardcode_into_libs=yes ;; freebsd3.[2-9]* | freebsdelf3.[2-9]* | \ freebsd4.[0-5] | freebsdelf4.[0-5] | freebsd4.1.1 | freebsdelf4.1.1) shlibpath_overrides_runpath=no hardcode_into_libs=yes ;; *) # from 4.6 on, and DragonFly shlibpath_overrides_runpath=yes hardcode_into_libs=yes ;; esac ;; haiku*) version_type=linux # correct to gnu/linux during the next big refactor need_lib_prefix=no need_version=no dynamic_linker="$host_os runtime_loader" library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' shlibpath_var=LIBRARY_PATH shlibpath_overrides_runpath=no sys_lib_dlsearch_path_spec='/boot/home/config/lib /boot/common/lib /boot/system/lib' hardcode_into_libs=yes ;; hpux9* | hpux10* | hpux11*) # Give a soname corresponding to the major version so that dld.sl refuses to # link against other versions. version_type=sunos need_lib_prefix=no need_version=no case $host_cpu in ia64*) shrext_cmds='.so' hardcode_into_libs=yes dynamic_linker="$host_os dld.so" shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=yes # Unless +noenvvar is specified. library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' if test 32 = "$HPUX_IA64_MODE"; then sys_lib_search_path_spec="/usr/lib/hpux32 /usr/local/lib/hpux32 /usr/local/lib" sys_lib_dlsearch_path_spec=/usr/lib/hpux32 else sys_lib_search_path_spec="/usr/lib/hpux64 /usr/local/lib/hpux64" sys_lib_dlsearch_path_spec=/usr/lib/hpux64 fi ;; hppa*64*) shrext_cmds='.sl' hardcode_into_libs=yes dynamic_linker="$host_os dld.sl" shlibpath_var=LD_LIBRARY_PATH # How should we handle SHLIB_PATH shlibpath_overrides_runpath=yes # Unless +noenvvar is specified. library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' sys_lib_search_path_spec="/usr/lib/pa20_64 /usr/ccs/lib/pa20_64" sys_lib_dlsearch_path_spec=$sys_lib_search_path_spec ;; *) shrext_cmds='.sl' dynamic_linker="$host_os dld.sl" shlibpath_var=SHLIB_PATH shlibpath_overrides_runpath=no # +s is required to enable SHLIB_PATH library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' ;; esac # HP-UX runs *really* slowly unless shared libraries are mode 555, ... postinstall_cmds='chmod 555 $lib' # or fails outright, so override atomically: install_override_mode=555 ;; interix[3-9]*) version_type=linux # correct to gnu/linux during the next big refactor need_lib_prefix=no need_version=no library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' dynamic_linker='Interix 3.x ld.so.1 (PE, like ELF)' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=no hardcode_into_libs=yes ;; irix5* | irix6* | nonstopux*) case $host_os in nonstopux*) version_type=nonstopux ;; *) if test yes = "$lt_cv_prog_gnu_ld"; then version_type=linux # correct to gnu/linux during the next big refactor else version_type=irix fi ;; esac need_lib_prefix=no need_version=no soname_spec='$libname$release$shared_ext$major' library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$release$shared_ext $libname$shared_ext' case $host_os in irix5* | nonstopux*) libsuff= shlibsuff= ;; *) case $LD in # libtool.m4 will add one of these switches to LD *-32|*"-32 "|*-melf32bsmip|*"-melf32bsmip ") libsuff= shlibsuff= libmagic=32-bit;; *-n32|*"-n32 "|*-melf32bmipn32|*"-melf32bmipn32 ") libsuff=32 shlibsuff=N32 libmagic=N32;; *-64|*"-64 "|*-melf64bmip|*"-melf64bmip ") libsuff=64 shlibsuff=64 libmagic=64-bit;; *) libsuff= shlibsuff= libmagic=never-match;; esac ;; esac shlibpath_var=LD_LIBRARY${shlibsuff}_PATH shlibpath_overrides_runpath=no sys_lib_search_path_spec="/usr/lib$libsuff /lib$libsuff /usr/local/lib$libsuff" sys_lib_dlsearch_path_spec="/usr/lib$libsuff /lib$libsuff" hardcode_into_libs=yes ;; # No shared lib support for Linux oldld, aout, or coff. linux*oldld* | linux*aout* | linux*coff*) dynamic_linker=no ;; linux*android*) version_type=none # Android doesn't support versioned libraries. need_lib_prefix=no need_version=no library_names_spec='$libname$release$shared_ext' soname_spec='$libname$release$shared_ext' finish_cmds= shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=yes # This implies no fast_install, which is unacceptable. # Some rework will be needed to allow for fast_install # before this can be enabled. hardcode_into_libs=yes dynamic_linker='Android linker' # Don't embed -rpath directories since the linker doesn't support them. hardcode_libdir_flag_spec='-L$libdir' ;; # This must be glibc/ELF. linux* | k*bsd*-gnu | kopensolaris*-gnu | gnu*) version_type=linux # correct to gnu/linux during the next big refactor need_lib_prefix=no need_version=no library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' finish_cmds='PATH="\$PATH:/sbin" ldconfig -n $libdir' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=no # Some binutils ld are patched to set DT_RUNPATH if ${lt_cv_shlibpath_overrides_runpath+:} false; then : $as_echo_n "(cached) " >&6 else lt_cv_shlibpath_overrides_runpath=no save_LDFLAGS=$LDFLAGS save_libdir=$libdir eval "libdir=/foo; wl=\"$lt_prog_compiler_wl\"; \ LDFLAGS=\"\$LDFLAGS $hardcode_libdir_flag_spec\"" cat confdefs.h - <<_ACEOF >conftest.$ac_ext /* end confdefs.h. */ int main () { ; return 0; } _ACEOF if ac_fn_c_try_link "$LINENO"; then : if ($OBJDUMP -p conftest$ac_exeext) 2>/dev/null | grep "RUNPATH.*$libdir" >/dev/null; then : lt_cv_shlibpath_overrides_runpath=yes fi fi rm -f core conftest.err conftest.$ac_objext \ conftest$ac_exeext conftest.$ac_ext LDFLAGS=$save_LDFLAGS libdir=$save_libdir fi shlibpath_overrides_runpath=$lt_cv_shlibpath_overrides_runpath # This implies no fast_install, which is unacceptable. # Some rework will be needed to allow for fast_install # before this can be enabled. hardcode_into_libs=yes # Ideally, we could use ldconfig to report *all* directores which are # searched for libraries, however this is still not possible. Aside from not # being certain /sbin/ldconfig is available, command # 'ldconfig -N -X -v | grep ^/' on 64bit Fedora does not report /usr/lib64, # even though it is searched at run-time. Try to do the best guess by # appending ld.so.conf contents (and includes) to the search path. if test -f /etc/ld.so.conf; then lt_ld_extra=`awk '/^include / { system(sprintf("cd /etc; cat %s 2>/dev/null", \$2)); skip = 1; } { if (!skip) print \$0; skip = 0; }' < /etc/ld.so.conf | $SED -e 's/#.*//;/^[ ]*hwcap[ ]/d;s/[:, ]/ /g;s/=[^=]*$//;s/=[^= ]* / /g;s/"//g;/^$/d' | tr '\n' ' '` sys_lib_dlsearch_path_spec="/lib /usr/lib $lt_ld_extra" fi # We used to test for /lib/ld.so.1 and disable shared libraries on # powerpc, because MkLinux only supported shared libraries with the # GNU dynamic linker. Since this was broken with cross compilers, # most powerpc-linux boxes support dynamic linking these days and # people can always --disable-shared, the test was removed, and we # assume the GNU/Linux dynamic linker is in use. dynamic_linker='GNU/Linux ld.so' ;; netbsd*) version_type=sunos need_lib_prefix=no need_version=no if echo __ELF__ | $CC -E - | $GREP __ELF__ >/dev/null; then library_names_spec='$libname$release$shared_ext$versuffix $libname$shared_ext$versuffix' finish_cmds='PATH="\$PATH:/sbin" ldconfig -m $libdir' dynamic_linker='NetBSD (a.out) ld.so' else library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' dynamic_linker='NetBSD ld.elf_so' fi shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=yes hardcode_into_libs=yes ;; newsos6) version_type=linux # correct to gnu/linux during the next big refactor library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=yes ;; *nto* | *qnx*) version_type=qnx need_lib_prefix=no need_version=no library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=no hardcode_into_libs=yes dynamic_linker='ldqnx.so' ;; openbsd* | bitrig*) version_type=sunos sys_lib_dlsearch_path_spec=/usr/lib need_lib_prefix=no if test -z "`echo __ELF__ | $CC -E - | $GREP __ELF__`"; then need_version=no else need_version=yes fi library_names_spec='$libname$release$shared_ext$versuffix $libname$shared_ext$versuffix' finish_cmds='PATH="\$PATH:/sbin" ldconfig -m $libdir' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=yes ;; os2*) libname_spec='$name' version_type=windows shrext_cmds=.dll need_version=no need_lib_prefix=no # OS/2 can only load a DLL with a base name of 8 characters or less. soname_spec='`test -n "$os2dllname" && libname="$os2dllname"; v=$($ECHO $release$versuffix | tr -d .-); n=$($ECHO $libname | cut -b -$((8 - ${#v})) | tr . _); $ECHO $n$v`$shared_ext' library_names_spec='${libname}_dll.$libext' dynamic_linker='OS/2 ld.exe' shlibpath_var=BEGINLIBPATH sys_lib_search_path_spec="/lib /usr/lib /usr/local/lib" sys_lib_dlsearch_path_spec=$sys_lib_search_path_spec postinstall_cmds='base_file=`basename \$file`~ dlpath=`$SHELL 2>&1 -c '\''. $dir/'\''\$base_file'\''i; $ECHO \$dlname'\''`~ dldir=$destdir/`dirname \$dlpath`~ test -d \$dldir || mkdir -p \$dldir~ $install_prog $dir/$dlname \$dldir/$dlname~ chmod a+x \$dldir/$dlname~ if test -n '\''$stripme'\'' && test -n '\''$striplib'\''; then eval '\''$striplib \$dldir/$dlname'\'' || exit \$?; fi' postuninstall_cmds='dldll=`$SHELL 2>&1 -c '\''. $file; $ECHO \$dlname'\''`~ dlpath=$dir/\$dldll~ $RM \$dlpath' ;; osf3* | osf4* | osf5*) version_type=osf need_lib_prefix=no need_version=no soname_spec='$libname$release$shared_ext$major' library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' shlibpath_var=LD_LIBRARY_PATH sys_lib_search_path_spec="/usr/shlib /usr/ccs/lib /usr/lib/cmplrs/cc /usr/lib /usr/local/lib /var/shlib" sys_lib_dlsearch_path_spec=$sys_lib_search_path_spec ;; rdos*) dynamic_linker=no ;; solaris*) version_type=linux # correct to gnu/linux during the next big refactor need_lib_prefix=no need_version=no library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=yes hardcode_into_libs=yes # ldd complains unless libraries are executable postinstall_cmds='chmod +x $lib' ;; sunos4*) version_type=sunos library_names_spec='$libname$release$shared_ext$versuffix $libname$shared_ext$versuffix' finish_cmds='PATH="\$PATH:/usr/etc" ldconfig $libdir' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=yes if test yes = "$with_gnu_ld"; then need_lib_prefix=no fi need_version=yes ;; sysv4 | sysv4.3*) version_type=linux # correct to gnu/linux during the next big refactor library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' shlibpath_var=LD_LIBRARY_PATH case $host_vendor in sni) shlibpath_overrides_runpath=no need_lib_prefix=no runpath_var=LD_RUN_PATH ;; siemens) need_lib_prefix=no ;; motorola) need_lib_prefix=no need_version=no shlibpath_overrides_runpath=no sys_lib_search_path_spec='/lib /usr/lib /usr/ccs/lib' ;; esac ;; sysv4*MP*) if test -d /usr/nec; then version_type=linux # correct to gnu/linux during the next big refactor library_names_spec='$libname$shared_ext.$versuffix $libname$shared_ext.$major $libname$shared_ext' soname_spec='$libname$shared_ext.$major' shlibpath_var=LD_LIBRARY_PATH fi ;; sysv5* | sco3.2v5* | sco5v6* | unixware* | OpenUNIX* | sysv4*uw2*) version_type=sco need_lib_prefix=no need_version=no library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=yes hardcode_into_libs=yes if test yes = "$with_gnu_ld"; then sys_lib_search_path_spec='/usr/local/lib /usr/gnu/lib /usr/ccs/lib /usr/lib /lib' else sys_lib_search_path_spec='/usr/ccs/lib /usr/lib' case $host_os in sco3.2v5*) sys_lib_search_path_spec="$sys_lib_search_path_spec /lib" ;; esac fi sys_lib_dlsearch_path_spec='/usr/lib' ;; tpf*) # TPF is a cross-target only. Preferred cross-host = GNU/Linux. version_type=linux # correct to gnu/linux during the next big refactor need_lib_prefix=no need_version=no library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' shlibpath_var=LD_LIBRARY_PATH shlibpath_overrides_runpath=no hardcode_into_libs=yes ;; uts4*) version_type=linux # correct to gnu/linux during the next big refactor library_names_spec='$libname$release$shared_ext$versuffix $libname$release$shared_ext$major $libname$shared_ext' soname_spec='$libname$release$shared_ext$major' shlibpath_var=LD_LIBRARY_PATH ;; *) dynamic_linker=no ;; esac { $as_echo "$as_me:${as_lineno-$LINENO}: result: $dynamic_linker" >&5 $as_echo "$dynamic_linker" >&6; } test no = "$dynamic_linker" && can_build_shared=no variables_saved_for_relink="PATH $shlibpath_var $runpath_var" if test yes = "$GCC"; then variables_saved_for_relink="$variables_saved_for_relink GCC_EXEC_PREFIX COMPILER_PATH LIBRARY_PATH" fi if test set = "${lt_cv_sys_lib_search_path_spec+set}"; then sys_lib_search_path_spec=$lt_cv_sys_lib_search_path_spec fi if test set = "${lt_cv_sys_lib_dlsearch_path_spec+set}"; then sys_lib_dlsearch_path_spec=$lt_cv_sys_lib_dlsearch_path_spec fi # remember unaugmented sys_lib_dlsearch_path content for libtool script decls... configure_time_dlsearch_path=$sys_lib_dlsearch_path_spec # ... but it needs LT_SYS_LIBRARY_PATH munging for other configure-time code func_munge_path_list sys_lib_dlsearch_path_spec "$LT_SYS_LIBRARY_PATH" # to be used as default LT_SYS_LIBRARY_PATH value in generated libtool configure_time_lt_sys_library_path=$LT_SYS_LIBRARY_PATH { $as_echo "$as_me:${as_lineno-$LINENO}: checking how to hardcode library paths into programs" >&5 $as_echo_n "checking how to hardcode library paths into programs... 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*-apple*) # Look for Core flag AC_ARG_WITH(core, [ --with-core = choose CoreAudio API support (mac only)]) AS_IF([test "x$with_core" == "xyes"], [ api="$api -D__MACOSX_CORE__" AC_MSG_RESULT(using CoreAudio) AC_CHECK_HEADER(CoreAudio/CoreAudio.h, [], [AC_MSG_ERROR(CoreAudio header files not found!)] ) LIBS="$LIBS -framework CoreAudio -framework CoreFoundation" ]) # If no audio api flags specified, use CoreAudio if [test "$api" == ""; ] then AC_SUBST( api, [-D__MACOSX_CORE__] ) AC_MSG_RESULT(using CoreAudio) AC_CHECK_HEADER(CoreAudio/CoreAudio.h, [], [AC_MSG_ERROR(CoreAudio header files not found!)] ) AC_SUBST( LIBS, ["-framework CoreAudio -framework CoreFoundation"] ) fi AC_CHECK_LIB(pthread, pthread_create, , AC_MSG_ERROR(RtAudio requires the pthread library!)) ;; *-mingw32*) AC_ARG_WITH(asio, [ --with-asio = choose ASIO API support (windoze only)]) AS_IF([test "x$with_asio" == "xyes"], [ api="$api -D__WINDOWS_ASIO__" AC_MSG_RESULT(using ASIO) AC_SUBST( objects, ["asio.o asiodrivers.o asiolist.o iasiothiscallresolver.o"] ) ]) # Look for DirectSound flag AC_ARG_WITH(ds, [ --with-ds = choose DirectSound API support (windoze only)]) AS_IF([test "x$with_ds" == "xyes"], [ api="$api -D__WINDOWS_DS__" AC_MSG_RESULT(using DirectSound) LIBS="-ldsound -lwinmm $LIBS" ]) # Look for WASAPI flag AC_ARG_WITH(wasapi, [ --with-wasapi = choose Windows Audio Session API support (windoze only)]) AS_IF([test "x$with_wasapi" == "xyes"], [ api="$api -D__WINDOWS_WASAPI__" AC_MSG_RESULT(using WASAPI) LIBS="-lwinmm -luuid -lksuser $LIBS" ]) # If no audio api flags specified, use DS if [test "$api" == "";] then AC_SUBST( api, [-D__WINDOWS_DS__] ) AC_MSG_RESULT(using DirectSound) LIBS="-ldsound -lwinmm $LIBS" fi LIBS="-lole32 $LIBS" ;; *) # Default case for unknown realtime systems. AC_MSG_ERROR(Unknown system type for realtime support!) ;; esac CPPFLAGS="$CPPFLAGS $api" AC_OUTPUT rtaudio-5.0.0/doc/000755 000766 000024 00000000000 13151630575 014044 5ustar00garystaff000000 000000 rtaudio-5.0.0/install.txt000644 000766 000024 00000005040 13151627174 015506 0ustar00garystaff000000 000000 RtAudio - a set of C++ classes which provide a common API for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio, and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound, ASIO and WASAPI) operating systems. By Gary P. Scavone, 2001-2017. To configure and compile (on Unix systems and MinGW): 1. Unpack the RtAudio distribution (tar -xzf rtaudio-x.x.tar.gz). 2. From within the directory containing this file, run configure: ./configure If you checked out the code from git, just run "autogen.sh". 3. Typing "make" will compile static and shared libraries, as well as the example programs in the "tests/" directory. A few options can be passed to configure (or the autogen.sh script), including: --enable-debug = enable various debug output --with-alsa = choose native ALSA API support (linux only) --with-pulse = choose native PulseAudio API support (linux only) --with-oss = choose OSS API support (unixes) --with-jack = choose JACK server support (linux or Macintosh OS-X) --with-core = choose CoreAudio API support (Macintosh OS-X only) --with-asio = choose ASIO API support (windows only) --with-wasapi = choose Windows Audio System API support (windows only) --with-ds = choose DirectSound API support (windows only) Typing "./configure --help" will display all the available options. Note that you can provide more than one "--with-" flag to the configure script to enable multiple API support. If you wish to use a different compiler than that selected by configure, specify that compiler in the command line (ex. to use CC): ./configure CXX=CC CMAKE USAGE: CMake support is provided via the CMakeLists.txt files. Assuming you have CMake installed on your system, a typical usage would involve the following steps (from within the parent distribution directory): mkdir _build_ cd _build_ cmake e.g. cmake .. -DAUDIO_WINDOWS_WASAPI=ON WINDOWS: All Windows audio APIs in RtAudio compile with either the MinGW compiler (tested with latest tdm64-gcc-4.8.1) or MS Visual Studio. Visual C++ 6.0 project files (very old) are included for the test programs in the /tests/Windows/ directory. These projects compile API support for ASIO, WASAPI and DirectSound. LINUX OSS: The OSS API support in RtAudio has not been tested for many years. I'm not even sure there are OSS drivers supporting recent linux kernels. 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By Gary P. Scavone, 2001-2017 (and many other developers!) This distribution of RtAudio contains the following: doc: RtAudio documentation (see doc/html/index.html) tests: example RtAudio programs include: header and source files necessary for ASIO, DS & OSS compilation tests/Windows: Visual C++ .net test program workspace and projects OVERVIEW: RtAudio is a set of C++ classes that provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio and OSS), Macintosh OS X and Windows (DirectSound, ASIO and WASAPI) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following objectives: - object-oriented C++ design - simple, common API across all supported platforms - only one source and one header file for easy inclusion in programming projects - allow simultaneous multi-api support - support dynamic connection of devices - provide extensive audio device parameter control - allow audio device capability probing - automatic internal conversion for data format, channel number compensation, (de)interleaving, and byte-swapping RtAudio incorporates the concept of audio streams, which represent audio output (playback) and/or input (recording). Available audio devices and their capabilities can be enumerated and then specified when opening a stream. Where applicable, multiple API support can be compiled and a particular API specified when creating an RtAudio instance. See the \ref apinotes section for information specific to each of the supported audio APIs. FURTHER READING: For complete documentation on RtAudio, see the doc directory of the distribution or surf to http://www.music.mcgill.ca/~gary/rtaudio/. LEGAL AND ETHICAL: The RtAudio license is similar to the MIT License. RtAudio: a set of realtime audio i/o C++ classes Copyright (c) 2001-2017 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions: The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software. Any person wishing to distribute modifications to the Software is asked to send the modifications to the original developer so that they can be incorporated into the canonical version. This is, however, not a binding provision of this license. THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. rtaudio-5.0.0/RtAudio.cpp000644 000766 000024 00001306415 13151627174 015365 0ustar00garystaff000000 000000 /************************************************************************/ /*! \class RtAudio \brief Realtime audio i/o C++ classes. RtAudio provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, Jack, and OSS), Macintosh OS X (CoreAudio and Jack), and Windows (DirectSound, ASIO and WASAPI) operating systems. RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ RtAudio: realtime audio i/o C++ classes Copyright (c) 2001-2017 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions: The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software. Any person wishing to distribute modifications to the Software is asked to send the modifications to the original developer so that they can be incorporated into the canonical version. This is, however, not a binding provision of this license. THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */ /************************************************************************/ // RtAudio: Version 5.0.0 #include "RtAudio.h" #include #include #include #include #include #include // Static variable definitions. const unsigned int RtApi::MAX_SAMPLE_RATES = 14; const unsigned int RtApi::SAMPLE_RATES[] = { 4000, 5512, 8000, 9600, 11025, 16000, 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000 }; #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) #define MUTEX_DESTROY(A) DeleteCriticalSection(A) #define MUTEX_LOCK(A) EnterCriticalSection(A) #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) #include "tchar.h" static std::string convertCharPointerToStdString(const char *text) { return std::string(text); } static std::string convertCharPointerToStdString(const wchar_t *text) { int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); std::string s( length-1, '\0' ); WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); return s; } #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) // pthread API #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) #define MUTEX_LOCK(A) pthread_mutex_lock(A) #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) #else #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions #define MUTEX_DESTROY(A) abs(*A) // dummy definitions #endif // *************************************************** // // // RtAudio definitions. // // *************************************************** // std::string RtAudio :: getVersion( void ) { return RTAUDIO_VERSION; } void RtAudio :: getCompiledApi( std::vector &apis ) { apis.clear(); // The order here will control the order of RtAudio's API search in // the constructor. #if defined(__UNIX_JACK__) apis.push_back( UNIX_JACK ); #endif #if defined(__LINUX_ALSA__) apis.push_back( LINUX_ALSA ); #endif #if defined(__LINUX_PULSE__) apis.push_back( LINUX_PULSE ); #endif #if defined(__LINUX_OSS__) apis.push_back( LINUX_OSS ); #endif #if defined(__WINDOWS_ASIO__) apis.push_back( WINDOWS_ASIO ); #endif #if defined(__WINDOWS_WASAPI__) apis.push_back( WINDOWS_WASAPI ); #endif #if defined(__WINDOWS_DS__) apis.push_back( WINDOWS_DS ); #endif #if defined(__MACOSX_CORE__) apis.push_back( MACOSX_CORE ); #endif #if defined(__RTAUDIO_DUMMY__) apis.push_back( RTAUDIO_DUMMY ); #endif } void RtAudio :: openRtApi( RtAudio::Api api ) { if ( rtapi_ ) delete rtapi_; rtapi_ = 0; #if defined(__UNIX_JACK__) if ( api == UNIX_JACK ) rtapi_ = new RtApiJack(); #endif #if defined(__LINUX_ALSA__) if ( api == LINUX_ALSA ) rtapi_ = new RtApiAlsa(); #endif #if defined(__LINUX_PULSE__) if ( api == LINUX_PULSE ) rtapi_ = new RtApiPulse(); #endif #if defined(__LINUX_OSS__) if ( api == LINUX_OSS ) rtapi_ = new RtApiOss(); #endif #if defined(__WINDOWS_ASIO__) if ( api == WINDOWS_ASIO ) rtapi_ = new RtApiAsio(); #endif #if defined(__WINDOWS_WASAPI__) if ( api == WINDOWS_WASAPI ) rtapi_ = new RtApiWasapi(); #endif #if defined(__WINDOWS_DS__) if ( api == WINDOWS_DS ) rtapi_ = new RtApiDs(); #endif #if defined(__MACOSX_CORE__) if ( api == MACOSX_CORE ) rtapi_ = new RtApiCore(); #endif #if defined(__RTAUDIO_DUMMY__) if ( api == RTAUDIO_DUMMY ) rtapi_ = new RtApiDummy(); #endif } RtAudio :: RtAudio( RtAudio::Api api ) { rtapi_ = 0; if ( api != UNSPECIFIED ) { // Attempt to open the specified API. openRtApi( api ); if ( rtapi_ ) return; // No compiled support for specified API value. Issue a debug // warning and continue as if no API was specified. std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; } // Iterate through the compiled APIs and return as soon as we find // one with at least one device or we reach the end of the list. std::vector< RtAudio::Api > apis; getCompiledApi( apis ); for ( unsigned int i=0; igetDeviceCount() ) break; } if ( rtapi_ ) return; // It should not be possible to get here because the preprocessor // definition __RTAUDIO_DUMMY__ is automatically defined if no // API-specific definitions are passed to the compiler. But just in // case something weird happens, we'll thow an error. std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n"; throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) ); } RtAudio :: ~RtAudio() { if ( rtapi_ ) delete rtapi_; } void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, RtAudio::StreamParameters *inputParameters, RtAudioFormat format, unsigned int sampleRate, unsigned int *bufferFrames, RtAudioCallback callback, void *userData, RtAudio::StreamOptions *options, RtAudioErrorCallback errorCallback ) { return rtapi_->openStream( outputParameters, inputParameters, format, sampleRate, bufferFrames, callback, userData, options, errorCallback ); } // *************************************************** // // // Public RtApi definitions (see end of file for // private or protected utility functions). // // *************************************************** // RtApi :: RtApi() { stream_.state = STREAM_CLOSED; stream_.mode = UNINITIALIZED; stream_.apiHandle = 0; stream_.userBuffer[0] = 0; stream_.userBuffer[1] = 0; MUTEX_INITIALIZE( &stream_.mutex ); showWarnings_ = true; firstErrorOccurred_ = false; } RtApi :: ~RtApi() { MUTEX_DESTROY( &stream_.mutex ); } void RtApi :: openStream( RtAudio::StreamParameters *oParams, RtAudio::StreamParameters *iParams, RtAudioFormat format, unsigned int sampleRate, unsigned int *bufferFrames, RtAudioCallback callback, void *userData, RtAudio::StreamOptions *options, RtAudioErrorCallback errorCallback ) { if ( stream_.state != STREAM_CLOSED ) { errorText_ = "RtApi::openStream: a stream is already open!"; error( RtAudioError::INVALID_USE ); return; } // Clear stream information potentially left from a previously open stream. clearStreamInfo(); if ( oParams && oParams->nChannels < 1 ) { errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; error( RtAudioError::INVALID_USE ); return; } if ( iParams && iParams->nChannels < 1 ) { errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; error( RtAudioError::INVALID_USE ); return; } if ( oParams == NULL && iParams == NULL ) { errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; error( RtAudioError::INVALID_USE ); return; } if ( formatBytes(format) == 0 ) { errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; error( RtAudioError::INVALID_USE ); return; } unsigned int nDevices = getDeviceCount(); unsigned int oChannels = 0; if ( oParams ) { oChannels = oParams->nChannels; if ( oParams->deviceId >= nDevices ) { errorText_ = "RtApi::openStream: output device parameter value is invalid."; error( RtAudioError::INVALID_USE ); return; } } unsigned int iChannels = 0; if ( iParams ) { iChannels = iParams->nChannels; if ( iParams->deviceId >= nDevices ) { errorText_ = "RtApi::openStream: input device parameter value is invalid."; error( RtAudioError::INVALID_USE ); return; } } bool result; if ( oChannels > 0 ) { result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, sampleRate, format, bufferFrames, options ); if ( result == false ) { error( RtAudioError::SYSTEM_ERROR ); return; } } if ( iChannels > 0 ) { result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, sampleRate, format, bufferFrames, options ); if ( result == false ) { if ( oChannels > 0 ) closeStream(); error( RtAudioError::SYSTEM_ERROR ); return; } } stream_.callbackInfo.callback = (void *) callback; stream_.callbackInfo.userData = userData; stream_.callbackInfo.errorCallback = (void *) errorCallback; if ( options ) options->numberOfBuffers = stream_.nBuffers; stream_.state = STREAM_STOPPED; } unsigned int RtApi :: getDefaultInputDevice( void ) { // Should be implemented in subclasses if possible. return 0; } unsigned int RtApi :: getDefaultOutputDevice( void ) { // Should be implemented in subclasses if possible. return 0; } void RtApi :: closeStream( void ) { // MUST be implemented in subclasses! return; } bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, RtAudio::StreamOptions * /*options*/ ) { // MUST be implemented in subclasses! return FAILURE; } void RtApi :: tickStreamTime( void ) { // Subclasses that do not provide their own implementation of // getStreamTime should call this function once per buffer I/O to // provide basic stream time support. stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); #if defined( HAVE_GETTIMEOFDAY ) gettimeofday( &stream_.lastTickTimestamp, NULL ); #endif } long RtApi :: getStreamLatency( void ) { verifyStream(); long totalLatency = 0; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) totalLatency = stream_.latency[0]; if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) totalLatency += stream_.latency[1]; return totalLatency; } double RtApi :: getStreamTime( void ) { verifyStream(); #if defined( HAVE_GETTIMEOFDAY ) // Return a very accurate estimate of the stream time by // adding in the elapsed time since the last tick. struct timeval then; struct timeval now; if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) return stream_.streamTime; gettimeofday( &now, NULL ); then = stream_.lastTickTimestamp; return stream_.streamTime + ((now.tv_sec + 0.000001 * now.tv_usec) - (then.tv_sec + 0.000001 * then.tv_usec)); #else return stream_.streamTime; #endif } void RtApi :: setStreamTime( double time ) { verifyStream(); if ( time >= 0.0 ) stream_.streamTime = time; #if defined( HAVE_GETTIMEOFDAY ) gettimeofday( &stream_.lastTickTimestamp, NULL ); #endif } unsigned int RtApi :: getStreamSampleRate( void ) { verifyStream(); return stream_.sampleRate; } // *************************************************** // // // OS/API-specific methods. // // *************************************************** // #if defined(__MACOSX_CORE__) // The OS X CoreAudio API is designed to use a separate callback // procedure for each of its audio devices. A single RtAudio duplex // stream using two different devices is supported here, though it // cannot be guaranteed to always behave correctly because we cannot // synchronize these two callbacks. // // A property listener is installed for over/underrun information. // However, no functionality is currently provided to allow property // listeners to trigger user handlers because it is unclear what could // be done if a critical stream parameter (buffer size, sample rate, // device disconnect) notification arrived. The listeners entail // quite a bit of extra code and most likely, a user program wouldn't // be prepared for the result anyway. However, we do provide a flag // to the client callback function to inform of an over/underrun. // A structure to hold various information related to the CoreAudio API // implementation. struct CoreHandle { AudioDeviceID id[2]; // device ids #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) AudioDeviceIOProcID procId[2]; #endif UInt32 iStream[2]; // device stream index (or first if using multiple) UInt32 nStreams[2]; // number of streams to use bool xrun[2]; char *deviceBuffer; pthread_cond_t condition; int drainCounter; // Tracks callback counts when draining bool internalDrain; // Indicates if stop is initiated from callback or not. CoreHandle() :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } }; RtApiCore:: RtApiCore() { #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER ) // This is a largely undocumented but absolutely necessary // requirement starting with OS-X 10.6. If not called, queries and // updates to various audio device properties are not handled // correctly. CFRunLoopRef theRunLoop = NULL; AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); if ( result != noErr ) { errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; error( RtAudioError::WARNING ); } #endif } RtApiCore :: ~RtApiCore() { // The subclass destructor gets called before the base class // destructor, so close an existing stream before deallocating // apiDeviceId memory. if ( stream_.state != STREAM_CLOSED ) closeStream(); } unsigned int RtApiCore :: getDeviceCount( void ) { // Find out how many audio devices there are, if any. UInt32 dataSize; AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize ); if ( result != noErr ) { errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; error( RtAudioError::WARNING ); return 0; } return dataSize / sizeof( AudioDeviceID ); } unsigned int RtApiCore :: getDefaultInputDevice( void ) { unsigned int nDevices = getDeviceCount(); if ( nDevices <= 1 ) return 0; AudioDeviceID id; UInt32 dataSize = sizeof( AudioDeviceID ); AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); if ( result != noErr ) { errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; error( RtAudioError::WARNING ); return 0; } dataSize *= nDevices; AudioDeviceID deviceList[ nDevices ]; property.mSelector = kAudioHardwarePropertyDevices; result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); if ( result != noErr ) { errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; error( RtAudioError::WARNING ); return 0; } for ( unsigned int i=0; i= nDevices ) { errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; error( RtAudioError::INVALID_USE ); return info; } AudioDeviceID deviceList[ nDevices ]; UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); if ( result != noErr ) { errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; error( RtAudioError::WARNING ); return info; } AudioDeviceID id = deviceList[ device ]; // Get the device name. info.name.erase(); CFStringRef cfname; dataSize = sizeof( CFStringRef ); property.mSelector = kAudioObjectPropertyManufacturer; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); int length = CFStringGetLength(cfname); char *mname = (char *)malloc(length * 3 + 1); #if defined( UNICODE ) || defined( _UNICODE ) CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8); #else CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); #endif info.name.append( (const char *)mname, strlen(mname) ); info.name.append( ": " ); CFRelease( cfname ); free(mname); property.mSelector = kAudioObjectPropertyName; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); length = CFStringGetLength(cfname); char *name = (char *)malloc(length * 3 + 1); #if defined( UNICODE ) || defined( _UNICODE ) CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8); #else CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); #endif info.name.append( (const char *)name, strlen(name) ); CFRelease( cfname ); free(name); // Get the output stream "configuration". AudioBufferList *bufferList = nil; property.mSelector = kAudioDevicePropertyStreamConfiguration; property.mScope = kAudioDevicePropertyScopeOutput; // property.mElement = kAudioObjectPropertyElementWildcard; dataSize = 0; result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); if ( result != noErr || dataSize == 0 ) { errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Allocate the AudioBufferList. bufferList = (AudioBufferList *) malloc( dataSize ); if ( bufferList == NULL ) { errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; error( RtAudioError::WARNING ); return info; } result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); if ( result != noErr || dataSize == 0 ) { free( bufferList ); errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Get output channel information. unsigned int i, nStreams = bufferList->mNumberBuffers; for ( i=0; imBuffers[i].mNumberChannels; free( bufferList ); // Get the input stream "configuration". property.mScope = kAudioDevicePropertyScopeInput; result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); if ( result != noErr || dataSize == 0 ) { errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Allocate the AudioBufferList. bufferList = (AudioBufferList *) malloc( dataSize ); if ( bufferList == NULL ) { errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; error( RtAudioError::WARNING ); return info; } result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); if (result != noErr || dataSize == 0) { free( bufferList ); errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Get input channel information. nStreams = bufferList->mNumberBuffers; for ( i=0; imBuffers[i].mNumberChannels; free( bufferList ); // If device opens for both playback and capture, we determine the channels. if ( info.outputChannels > 0 && info.inputChannels > 0 ) info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; // Probe the device sample rates. bool isInput = false; if ( info.outputChannels == 0 ) isInput = true; // Determine the supported sample rates. property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput; result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); if ( result != kAudioHardwareNoError || dataSize == 0 ) { errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } UInt32 nRanges = dataSize / sizeof( AudioValueRange ); AudioValueRange rangeList[ nRanges ]; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList ); if ( result != kAudioHardwareNoError ) { errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // The sample rate reporting mechanism is a bit of a mystery. It // seems that it can either return individual rates or a range of // rates. I assume that if the min / max range values are the same, // then that represents a single supported rate and if the min / max // range values are different, the device supports an arbitrary // range of values (though there might be multiple ranges, so we'll // use the most conservative range). Float64 minimumRate = 1.0, maximumRate = 10000000000.0; bool haveValueRange = false; info.sampleRates.clear(); for ( UInt32 i=0; i info.preferredSampleRate ) ) info.preferredSampleRate = tmpSr; } else { haveValueRange = true; if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum; if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum; } } if ( haveValueRange ) { for ( unsigned int k=0; k= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) { info.sampleRates.push_back( SAMPLE_RATES[k] ); if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) info.preferredSampleRate = SAMPLE_RATES[k]; } } } // Sort and remove any redundant values std::sort( info.sampleRates.begin(), info.sampleRates.end() ); info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() ); if ( info.sampleRates.size() == 0 ) { errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // CoreAudio always uses 32-bit floating point data for PCM streams. // Thus, any other "physical" formats supported by the device are of // no interest to the client. info.nativeFormats = RTAUDIO_FLOAT32; if ( info.outputChannels > 0 ) if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; if ( info.inputChannels > 0 ) if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; info.probed = true; return info; } static OSStatus callbackHandler( AudioDeviceID inDevice, const AudioTimeStamp* /*inNow*/, const AudioBufferList* inInputData, const AudioTimeStamp* /*inInputTime*/, AudioBufferList* outOutputData, const AudioTimeStamp* /*inOutputTime*/, void* infoPointer ) { CallbackInfo *info = (CallbackInfo *) infoPointer; RtApiCore *object = (RtApiCore *) info->object; if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) return kAudioHardwareUnspecifiedError; else return kAudioHardwareNoError; } static OSStatus xrunListener( AudioObjectID /*inDevice*/, UInt32 nAddresses, const AudioObjectPropertyAddress properties[], void* handlePointer ) { CoreHandle *handle = (CoreHandle *) handlePointer; for ( UInt32 i=0; ixrun[1] = true; else handle->xrun[0] = true; } } return kAudioHardwareNoError; } static OSStatus rateListener( AudioObjectID inDevice, UInt32 /*nAddresses*/, const AudioObjectPropertyAddress /*properties*/[], void* ratePointer ) { Float64 *rate = (Float64 *) ratePointer; UInt32 dataSize = sizeof( Float64 ); AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate ); return kAudioHardwareNoError; } bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ) { // Get device ID unsigned int nDevices = getDeviceCount(); if ( nDevices == 0 ) { // This should not happen because a check is made before this function is called. errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; return FAILURE; } if ( device >= nDevices ) { // This should not happen because a check is made before this function is called. errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; return FAILURE; } AudioDeviceID deviceList[ nDevices ]; UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); if ( result != noErr ) { errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; return FAILURE; } AudioDeviceID id = deviceList[ device ]; // Setup for stream mode. bool isInput = false; if ( mode == INPUT ) { isInput = true; property.mScope = kAudioDevicePropertyScopeInput; } else property.mScope = kAudioDevicePropertyScopeOutput; // Get the stream "configuration". AudioBufferList *bufferList = nil; dataSize = 0; property.mSelector = kAudioDevicePropertyStreamConfiguration; result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); if ( result != noErr || dataSize == 0 ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } // Allocate the AudioBufferList. bufferList = (AudioBufferList *) malloc( dataSize ); if ( bufferList == NULL ) { errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; return FAILURE; } result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); if (result != noErr || dataSize == 0) { free( bufferList ); errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } // Search for one or more streams that contain the desired number of // channels. CoreAudio devices can have an arbitrary number of // streams and each stream can have an arbitrary number of channels. // For each stream, a single buffer of interleaved samples is // provided. RtAudio prefers the use of one stream of interleaved // data or multiple consecutive single-channel streams. However, we // now support multiple consecutive multi-channel streams of // interleaved data as well. UInt32 iStream, offsetCounter = firstChannel; UInt32 nStreams = bufferList->mNumberBuffers; bool monoMode = false; bool foundStream = false; // First check that the device supports the requested number of // channels. UInt32 deviceChannels = 0; for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; if ( deviceChannels < ( channels + firstChannel ) ) { free( bufferList ); errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; errorText_ = errorStream_.str(); return FAILURE; } // Look for a single stream meeting our needs. UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; if ( streamChannels >= channels + offsetCounter ) { firstStream = iStream; channelOffset = offsetCounter; foundStream = true; break; } if ( streamChannels > offsetCounter ) break; offsetCounter -= streamChannels; } // If we didn't find a single stream above, then we should be able // to meet the channel specification with multiple streams. if ( foundStream == false ) { monoMode = true; offsetCounter = firstChannel; for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; if ( streamChannels > offsetCounter ) break; offsetCounter -= streamChannels; } firstStream = iStream; channelOffset = offsetCounter; Int32 channelCounter = channels + offsetCounter - streamChannels; if ( streamChannels > 1 ) monoMode = false; while ( channelCounter > 0 ) { streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; if ( streamChannels > 1 ) monoMode = false; channelCounter -= streamChannels; streamCount++; } } free( bufferList ); // Determine the buffer size. AudioValueRange bufferRange; dataSize = sizeof( AudioValueRange ); property.mSelector = kAudioDevicePropertyBufferFrameSizeRange; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; // Set the buffer size. For multiple streams, I'm assuming we only // need to make this setting for the master channel. UInt32 theSize = (UInt32) *bufferSize; dataSize = sizeof( UInt32 ); property.mSelector = kAudioDevicePropertyBufferFrameSize; result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } // If attempting to setup a duplex stream, the bufferSize parameter // MUST be the same in both directions! *bufferSize = theSize; if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } stream_.bufferSize = *bufferSize; stream_.nBuffers = 1; // Try to set "hog" mode ... it's not clear to me this is working. if ( options && options->flags & RTAUDIO_HOG_DEVICE ) { pid_t hog_pid; dataSize = sizeof( hog_pid ); property.mSelector = kAudioDevicePropertyHogMode; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!"; errorText_ = errorStream_.str(); return FAILURE; } if ( hog_pid != getpid() ) { hog_pid = getpid(); result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; errorText_ = errorStream_.str(); return FAILURE; } } } // Check and if necessary, change the sample rate for the device. Float64 nominalRate; dataSize = sizeof( Float64 ); property.mSelector = kAudioDevicePropertyNominalSampleRate; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate."; errorText_ = errorStream_.str(); return FAILURE; } // Only change the sample rate if off by more than 1 Hz. if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) { // Set a property listener for the sample rate change Float64 reportedRate = 0.0; AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } nominalRate = (Float64) sampleRate; result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); if ( result != noErr ) { AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } // Now wait until the reported nominal rate is what we just set. UInt32 microCounter = 0; while ( reportedRate != nominalRate ) { microCounter += 5000; if ( microCounter > 5000000 ) break; usleep( 5000 ); } // Remove the property listener. AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); if ( microCounter > 5000000 ) { errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } } // Now set the stream format for all streams. Also, check the // physical format of the device and change that if necessary. AudioStreamBasicDescription description; dataSize = sizeof( AudioStreamBasicDescription ); property.mSelector = kAudioStreamPropertyVirtualFormat; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } // Set the sample rate and data format id. However, only make the // change if the sample rate is not within 1.0 of the desired // rate and the format is not linear pcm. bool updateFormat = false; if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) { description.mSampleRate = (Float64) sampleRate; updateFormat = true; } if ( description.mFormatID != kAudioFormatLinearPCM ) { description.mFormatID = kAudioFormatLinearPCM; updateFormat = true; } if ( updateFormat ) { result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } } // Now check the physical format. property.mSelector = kAudioStreamPropertyPhysicalFormat; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; } //std::cout << "Current physical stream format:" << std::endl; //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl; //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl; //std::cout << " sample rate = " << description.mSampleRate << std::endl; if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) { description.mFormatID = kAudioFormatLinearPCM; //description.mSampleRate = (Float64) sampleRate; AudioStreamBasicDescription testDescription = description; UInt32 formatFlags; // We'll try higher bit rates first and then work our way down. std::vector< std::pair > physicalFormats; formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; physicalFormats.push_back( std::pair( 32, formatFlags ) ); formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; physicalFormats.push_back( std::pair( 32, formatFlags ) ); physicalFormats.push_back( std::pair( 24, formatFlags ) ); // 24-bit packed formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh ); physicalFormats.push_back( std::pair( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low formatFlags |= kAudioFormatFlagIsAlignedHigh; physicalFormats.push_back( std::pair( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; physicalFormats.push_back( std::pair( 16, formatFlags ) ); physicalFormats.push_back( std::pair( 8, formatFlags ) ); bool setPhysicalFormat = false; for( unsigned int i=0; iflags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; else stream_.userInterleaved = true; stream_.deviceInterleaved[mode] = true; if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] ) stream_.doConvertBuffer[mode] = true; if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) stream_.doConvertBuffer[mode] = true; if ( streamCount == 1 ) { if ( stream_.nUserChannels[mode] > 1 && stream_.userInterleaved != stream_.deviceInterleaved[mode] ) stream_.doConvertBuffer[mode] = true; } else if ( monoMode && stream_.userInterleaved ) stream_.doConvertBuffer[mode] = true; // Allocate our CoreHandle structure for the stream. CoreHandle *handle = 0; if ( stream_.apiHandle == 0 ) { try { handle = new CoreHandle; } catch ( std::bad_alloc& ) { errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; goto error; } if ( pthread_cond_init( &handle->condition, NULL ) ) { errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; goto error; } stream_.apiHandle = (void *) handle; } else handle = (CoreHandle *) stream_.apiHandle; handle->iStream[mode] = firstStream; handle->nStreams[mode] = streamCount; handle->id[mode] = id; // Allocate necessary internal buffers. unsigned long bufferBytes; bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) ); memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; goto error; } // If possible, we will make use of the CoreAudio stream buffers as // "device buffers". However, we can't do this if using multiple // streams. if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); if ( mode == INPUT ) { if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); if ( bufferBytes <= bytesOut ) makeBuffer = false; } } if ( makeBuffer ) { bufferBytes *= *bufferSize; if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); if ( stream_.deviceBuffer == NULL ) { errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; goto error; } } } stream_.sampleRate = sampleRate; stream_.device[mode] = device; stream_.state = STREAM_STOPPED; stream_.callbackInfo.object = (void *) this; // Setup the buffer conversion information structure. if ( stream_.doConvertBuffer[mode] ) { if ( streamCount > 1 ) setConvertInfo( mode, 0 ); else setConvertInfo( mode, channelOffset ); } if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) // Only one callback procedure per device. stream_.mode = DUPLEX; else { #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); #else // deprecated in favor of AudioDeviceCreateIOProcID() result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); #endif if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; errorText_ = errorStream_.str(); goto error; } if ( stream_.mode == OUTPUT && mode == INPUT ) stream_.mode = DUPLEX; else stream_.mode = mode; } // Setup the device property listener for over/underload. property.mSelector = kAudioDeviceProcessorOverload; property.mScope = kAudioObjectPropertyScopeGlobal; result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); return SUCCESS; error: if ( handle ) { pthread_cond_destroy( &handle->condition ); delete handle; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } stream_.state = STREAM_CLOSED; return FAILURE; } void RtApiCore :: closeStream( void ) { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiCore::closeStream(): no open stream to close!"; error( RtAudioError::WARNING ); return; } CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if (handle) { AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; property.mSelector = kAudioDeviceProcessorOverload; property.mScope = kAudioObjectPropertyScopeGlobal; if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { errorText_ = "RtApiCore::closeStream(): error removing property listener!"; error( RtAudioError::WARNING ); } } if ( stream_.state == STREAM_RUNNING ) AudioDeviceStop( handle->id[0], callbackHandler ); #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); #else // deprecated in favor of AudioDeviceDestroyIOProcID() AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); #endif } if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { if (handle) { AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; property.mSelector = kAudioDeviceProcessorOverload; property.mScope = kAudioObjectPropertyScopeGlobal; if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) { errorText_ = "RtApiCore::closeStream(): error removing property listener!"; error( RtAudioError::WARNING ); } } if ( stream_.state == STREAM_RUNNING ) AudioDeviceStop( handle->id[1], callbackHandler ); #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); #else // deprecated in favor of AudioDeviceDestroyIOProcID() AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); #endif } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } // Destroy pthread condition variable. pthread_cond_destroy( &handle->condition ); delete handle; stream_.apiHandle = 0; stream_.mode = UNINITIALIZED; stream_.state = STREAM_CLOSED; } void RtApiCore :: startStream( void ) { verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiCore::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } OSStatus result = noErr; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { result = AudioDeviceStart( handle->id[0], callbackHandler ); if ( result != noErr ) { errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; errorText_ = errorStream_.str(); goto unlock; } } if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { result = AudioDeviceStart( handle->id[1], callbackHandler ); if ( result != noErr ) { errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; errorText_ = errorStream_.str(); goto unlock; } } handle->drainCounter = 0; handle->internalDrain = false; stream_.state = STREAM_RUNNING; unlock: if ( result == noErr ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiCore :: stopStream( void ) { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } OSStatus result = noErr; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( handle->drainCounter == 0 ) { handle->drainCounter = 2; pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled } result = AudioDeviceStop( handle->id[0], callbackHandler ); if ( result != noErr ) { errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ")."; errorText_ = errorStream_.str(); goto unlock; } } if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { result = AudioDeviceStop( handle->id[1], callbackHandler ); if ( result != noErr ) { errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ")."; errorText_ = errorStream_.str(); goto unlock; } } stream_.state = STREAM_STOPPED; unlock: if ( result == noErr ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiCore :: abortStream( void ) { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } CoreHandle *handle = (CoreHandle *) stream_.apiHandle; handle->drainCounter = 2; stopStream(); } // This function will be called by a spawned thread when the user // callback function signals that the stream should be stopped or // aborted. It is better to handle it this way because the // callbackEvent() function probably should return before the AudioDeviceStop() // function is called. static void *coreStopStream( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiCore *object = (RtApiCore *) info->object; object->stopStream(); pthread_exit( NULL ); } bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, const AudioBufferList *inBufferList, const AudioBufferList *outBufferList ) { if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; error( RtAudioError::WARNING ); return FAILURE; } CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > 3 ) { ThreadHandle threadId; stream_.state = STREAM_STOPPING; if ( handle->internalDrain == true ) pthread_create( &threadId, NULL, coreStopStream, info ); else // external call to stopStream() pthread_cond_signal( &handle->condition ); return SUCCESS; } AudioDeviceID outputDevice = handle->id[0]; // Invoke user callback to get fresh output data UNLESS we are // draining stream or duplex mode AND the input/output devices are // different AND this function is called for the input device. if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { RtAudioCallback callback = (RtAudioCallback) info->callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; if ( stream_.mode != INPUT && handle->xrun[0] == true ) { status |= RTAUDIO_OUTPUT_UNDERFLOW; handle->xrun[0] = false; } if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { status |= RTAUDIO_INPUT_OVERFLOW; handle->xrun[1] = false; } int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); if ( cbReturnValue == 2 ) { stream_.state = STREAM_STOPPING; handle->drainCounter = 2; abortStream(); return SUCCESS; } else if ( cbReturnValue == 1 ) { handle->drainCounter = 1; handle->internalDrain = true; } } if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { if ( handle->drainCounter > 1 ) { // write zeros to the output stream if ( handle->nStreams[0] == 1 ) { memset( outBufferList->mBuffers[handle->iStream[0]].mData, 0, outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); } else { // fill multiple streams with zeros for ( unsigned int i=0; inStreams[0]; i++ ) { memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, 0, outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); } } } else if ( handle->nStreams[0] == 1 ) { if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, stream_.userBuffer[0], stream_.convertInfo[0] ); } else { // copy from user buffer memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, stream_.userBuffer[0], outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); } } else { // fill multiple streams Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; if ( stream_.doConvertBuffer[0] ) { convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); inBuffer = (Float32 *) stream_.deviceBuffer; } if ( stream_.deviceInterleaved[0] == false ) { // mono mode UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); } } else { // fill multiple multi-channel streams with interleaved data UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; Float32 *out, *in; bool inInterleaved = ( stream_.userInterleaved ) ? true : false; UInt32 inChannels = stream_.nUserChannels[0]; if ( stream_.doConvertBuffer[0] ) { inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode inChannels = stream_.nDeviceChannels[0]; } if ( inInterleaved ) inOffset = 1; else inOffset = stream_.bufferSize; channelsLeft = inChannels; for ( unsigned int i=0; inStreams[0]; i++ ) { in = inBuffer; out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; outJump = 0; // Account for possible channel offset in first stream if ( i == 0 && stream_.channelOffset[0] > 0 ) { streamChannels -= stream_.channelOffset[0]; outJump = stream_.channelOffset[0]; out += outJump; } // Account for possible unfilled channels at end of the last stream if ( streamChannels > channelsLeft ) { outJump = streamChannels - channelsLeft; streamChannels = channelsLeft; } // Determine input buffer offsets and skips if ( inInterleaved ) { inJump = inChannels; in += inChannels - channelsLeft; } else { inJump = 1; in += (inChannels - channelsLeft) * inOffset; } for ( unsigned int i=0; idrainCounter ) { handle->drainCounter++; goto unlock; } AudioDeviceID inputDevice; inputDevice = handle->id[1]; if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { if ( handle->nStreams[1] == 1 ) { if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer convertBuffer( stream_.userBuffer[1], (char *) inBufferList->mBuffers[handle->iStream[1]].mData, stream_.convertInfo[1] ); } else { // copy to user buffer memcpy( stream_.userBuffer[1], inBufferList->mBuffers[handle->iStream[1]].mData, inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); } } else { // read from multiple streams Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; if ( stream_.deviceInterleaved[1] == false ) { // mono mode UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); } } else { // read from multiple multi-channel streams UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; Float32 *out, *in; bool outInterleaved = ( stream_.userInterleaved ) ? true : false; UInt32 outChannels = stream_.nUserChannels[1]; if ( stream_.doConvertBuffer[1] ) { outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode outChannels = stream_.nDeviceChannels[1]; } if ( outInterleaved ) outOffset = 1; else outOffset = stream_.bufferSize; channelsLeft = outChannels; for ( unsigned int i=0; inStreams[1]; i++ ) { out = outBuffer; in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; inJump = 0; // Account for possible channel offset in first stream if ( i == 0 && stream_.channelOffset[1] > 0 ) { streamChannels -= stream_.channelOffset[1]; inJump = stream_.channelOffset[1]; in += inJump; } // Account for possible unread channels at end of the last stream if ( streamChannels > channelsLeft ) { inJump = streamChannels - channelsLeft; streamChannels = channelsLeft; } // Determine output buffer offsets and skips if ( outInterleaved ) { outJump = outChannels; out += outChannels - channelsLeft; } else { outJump = 1; out += (outChannels - channelsLeft) * outOffset; } for ( unsigned int i=0; i #include #include // A structure to hold various information related to the Jack API // implementation. struct JackHandle { jack_client_t *client; jack_port_t **ports[2]; std::string deviceName[2]; bool xrun[2]; pthread_cond_t condition; int drainCounter; // Tracks callback counts when draining bool internalDrain; // Indicates if stop is initiated from callback or not. JackHandle() :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } }; #if !defined(__RTAUDIO_DEBUG__) static void jackSilentError( const char * ) {}; #endif RtApiJack :: RtApiJack() :shouldAutoconnect_(true) { // Nothing to do here. #if !defined(__RTAUDIO_DEBUG__) // Turn off Jack's internal error reporting. jack_set_error_function( &jackSilentError ); #endif } RtApiJack :: ~RtApiJack() { if ( stream_.state != STREAM_CLOSED ) closeStream(); } unsigned int RtApiJack :: getDeviceCount( void ) { // See if we can become a jack client. jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption; jack_status_t *status = NULL; jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); if ( client == 0 ) return 0; const char **ports; std::string port, previousPort; unsigned int nChannels = 0, nDevices = 0; ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). size_t iColon = 0; do { port = (char *) ports[ nChannels ]; iColon = port.find(":"); if ( iColon != std::string::npos ) { port = port.substr( 0, iColon + 1 ); if ( port != previousPort ) { nDevices++; previousPort = port; } } } while ( ports[++nChannels] ); free( ports ); } jack_client_close( client ); return nDevices; } RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; info.probed = false; jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption jack_status_t *status = NULL; jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); if ( client == 0 ) { errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; error( RtAudioError::WARNING ); return info; } const char **ports; std::string port, previousPort; unsigned int nPorts = 0, nDevices = 0; ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). size_t iColon = 0; do { port = (char *) ports[ nPorts ]; iColon = port.find(":"); if ( iColon != std::string::npos ) { port = port.substr( 0, iColon ); if ( port != previousPort ) { if ( nDevices == device ) info.name = port; nDevices++; previousPort = port; } } } while ( ports[++nPorts] ); free( ports ); } if ( device >= nDevices ) { jack_client_close( client ); errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; error( RtAudioError::INVALID_USE ); return info; } // Get the current jack server sample rate. info.sampleRates.clear(); info.preferredSampleRate = jack_get_sample_rate( client ); info.sampleRates.push_back( info.preferredSampleRate ); // Count the available ports containing the client name as device // channels. Jack "input ports" equal RtAudio output channels. unsigned int nChannels = 0; ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); if ( ports ) { while ( ports[ nChannels ] ) nChannels++; free( ports ); info.outputChannels = nChannels; } // Jack "output ports" equal RtAudio input channels. nChannels = 0; ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); if ( ports ) { while ( ports[ nChannels ] ) nChannels++; free( ports ); info.inputChannels = nChannels; } if ( info.outputChannels == 0 && info.inputChannels == 0 ) { jack_client_close(client); errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; error( RtAudioError::WARNING ); return info; } // If device opens for both playback and capture, we determine the channels. if ( info.outputChannels > 0 && info.inputChannels > 0 ) info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; // Jack always uses 32-bit floats. info.nativeFormats = RTAUDIO_FLOAT32; // Jack doesn't provide default devices so we'll use the first available one. if ( device == 0 && info.outputChannels > 0 ) info.isDefaultOutput = true; if ( device == 0 && info.inputChannels > 0 ) info.isDefaultInput = true; jack_client_close(client); info.probed = true; return info; } static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) { CallbackInfo *info = (CallbackInfo *) infoPointer; RtApiJack *object = (RtApiJack *) info->object; if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; return 0; } // This function will be called by a spawned thread when the Jack // server signals that it is shutting down. It is necessary to handle // it this way because the jackShutdown() function must return before // the jack_deactivate() function (in closeStream()) will return. static void *jackCloseStream( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiJack *object = (RtApiJack *) info->object; object->closeStream(); pthread_exit( NULL ); } static void jackShutdown( void *infoPointer ) { CallbackInfo *info = (CallbackInfo *) infoPointer; RtApiJack *object = (RtApiJack *) info->object; // Check current stream state. If stopped, then we'll assume this // was called as a result of a call to RtApiJack::stopStream (the // deactivation of a client handle causes this function to be called). // If not, we'll assume the Jack server is shutting down or some // other problem occurred and we should close the stream. if ( object->isStreamRunning() == false ) return; ThreadHandle threadId; pthread_create( &threadId, NULL, jackCloseStream, info ); std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; } static int jackXrun( void *infoPointer ) { JackHandle *handle = (JackHandle *) infoPointer; if ( handle->ports[0] ) handle->xrun[0] = true; if ( handle->ports[1] ) handle->xrun[1] = true; return 0; } bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ) { JackHandle *handle = (JackHandle *) stream_.apiHandle; // Look for jack server and try to become a client (only do once per stream). jack_client_t *client = 0; if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption; jack_status_t *status = NULL; if ( options && !options->streamName.empty() ) client = jack_client_open( options->streamName.c_str(), jackoptions, status ); else client = jack_client_open( "RtApiJack", jackoptions, status ); if ( client == 0 ) { errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; error( RtAudioError::WARNING ); return FAILURE; } } else { // The handle must have been created on an earlier pass. client = handle->client; } const char **ports; std::string port, previousPort, deviceName; unsigned int nPorts = 0, nDevices = 0; ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). size_t iColon = 0; do { port = (char *) ports[ nPorts ]; iColon = port.find(":"); if ( iColon != std::string::npos ) { port = port.substr( 0, iColon ); if ( port != previousPort ) { if ( nDevices == device ) deviceName = port; nDevices++; previousPort = port; } } } while ( ports[++nPorts] ); free( ports ); } if ( device >= nDevices ) { errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; return FAILURE; } // Count the available ports containing the client name as device // channels. Jack "input ports" equal RtAudio output channels. unsigned int nChannels = 0; unsigned long flag = JackPortIsInput; if ( mode == INPUT ) flag = JackPortIsOutput; ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); if ( ports ) { while ( ports[ nChannels ] ) nChannels++; free( ports ); } // Compare the jack ports for specified client to the requested number of channels. if ( nChannels < (channels + firstChannel) ) { errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; errorText_ = errorStream_.str(); return FAILURE; } // Check the jack server sample rate. unsigned int jackRate = jack_get_sample_rate( client ); if ( sampleRate != jackRate ) { jack_client_close( client ); errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; errorText_ = errorStream_.str(); return FAILURE; } stream_.sampleRate = jackRate; // Get the latency of the JACK port. ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); if ( ports[ firstChannel ] ) { // Added by Ge Wang jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); // the range (usually the min and max are equal) jack_latency_range_t latrange; latrange.min = latrange.max = 0; // get the latency range jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange ); // be optimistic, use the min! stream_.latency[mode] = latrange.min; //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); } free( ports ); // The jack server always uses 32-bit floating-point data. stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; stream_.userFormat = format; if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; else stream_.userInterleaved = true; // Jack always uses non-interleaved buffers. stream_.deviceInterleaved[mode] = false; // Jack always provides host byte-ordered data. stream_.doByteSwap[mode] = false; // Get the buffer size. The buffer size and number of buffers // (periods) is set when the jack server is started. stream_.bufferSize = (int) jack_get_buffer_size( client ); *bufferSize = stream_.bufferSize; stream_.nDeviceChannels[mode] = channels; stream_.nUserChannels[mode] = channels; // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] ) stream_.doConvertBuffer[mode] = true; if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && stream_.nUserChannels[mode] > 1 ) stream_.doConvertBuffer[mode] = true; // Allocate our JackHandle structure for the stream. if ( handle == 0 ) { try { handle = new JackHandle; } catch ( std::bad_alloc& ) { errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; goto error; } if ( pthread_cond_init(&handle->condition, NULL) ) { errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; goto error; } stream_.apiHandle = (void *) handle; handle->client = client; } handle->deviceName[mode] = deviceName; // Allocate necessary internal buffers. unsigned long bufferBytes; bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; goto error; } if ( stream_.doConvertBuffer[mode] ) { bool makeBuffer = true; if ( mode == OUTPUT ) bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); else { // mode == INPUT bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); if ( bufferBytes < bytesOut ) makeBuffer = false; } } if ( makeBuffer ) { bufferBytes *= *bufferSize; if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); if ( stream_.deviceBuffer == NULL ) { errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; goto error; } } } // Allocate memory for the Jack ports (channels) identifiers. handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); if ( handle->ports[mode] == NULL ) { errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; goto error; } stream_.device[mode] = device; stream_.channelOffset[mode] = firstChannel; stream_.state = STREAM_STOPPED; stream_.callbackInfo.object = (void *) this; if ( stream_.mode == OUTPUT && mode == INPUT ) // We had already set up the stream for output. stream_.mode = DUPLEX; else { stream_.mode = mode; jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); } // Register our ports. char label[64]; if ( mode == OUTPUT ) { for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); } } else { for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); } } // Setup the buffer conversion information structure. We don't use // buffers to do channel offsets, so we override that parameter // here. if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false; return SUCCESS; error: if ( handle ) { pthread_cond_destroy( &handle->condition ); jack_client_close( handle->client ); if ( handle->ports[0] ) free( handle->ports[0] ); if ( handle->ports[1] ) free( handle->ports[1] ); delete handle; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } return FAILURE; } void RtApiJack :: closeStream( void ) { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiJack::closeStream(): no open stream to close!"; error( RtAudioError::WARNING ); return; } JackHandle *handle = (JackHandle *) stream_.apiHandle; if ( handle ) { if ( stream_.state == STREAM_RUNNING ) jack_deactivate( handle->client ); jack_client_close( handle->client ); } if ( handle ) { if ( handle->ports[0] ) free( handle->ports[0] ); if ( handle->ports[1] ) free( handle->ports[1] ); pthread_cond_destroy( &handle->condition ); delete handle; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } stream_.mode = UNINITIALIZED; stream_.state = STREAM_CLOSED; } void RtApiJack :: startStream( void ) { verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiJack::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } JackHandle *handle = (JackHandle *) stream_.apiHandle; int result = jack_activate( handle->client ); if ( result ) { errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; goto unlock; } const char **ports; // Get the list of available ports. if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) { result = 1; ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); if ( ports == NULL) { errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; goto unlock; } // Now make the port connections. Since RtAudio wasn't designed to // allow the user to select particular channels of a device, we'll // just open the first "nChannels" ports with offset. for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); if ( result ) { free( ports ); errorText_ = "RtApiJack::startStream(): error connecting output ports!"; goto unlock; } } free(ports); } if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) { result = 1; ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); if ( ports == NULL) { errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; goto unlock; } // Now make the port connections. See note above. for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); if ( result ) { free( ports ); errorText_ = "RtApiJack::startStream(): error connecting input ports!"; goto unlock; } } free(ports); } handle->drainCounter = 0; handle->internalDrain = false; stream_.state = STREAM_RUNNING; unlock: if ( result == 0 ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiJack :: stopStream( void ) { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } JackHandle *handle = (JackHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( handle->drainCounter == 0 ) { handle->drainCounter = 2; pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled } } jack_deactivate( handle->client ); stream_.state = STREAM_STOPPED; } void RtApiJack :: abortStream( void ) { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } JackHandle *handle = (JackHandle *) stream_.apiHandle; handle->drainCounter = 2; stopStream(); } // This function will be called by a spawned thread when the user // callback function signals that the stream should be stopped or // aborted. It is necessary to handle it this way because the // callbackEvent() function must return before the jack_deactivate() // function will return. static void *jackStopStream( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiJack *object = (RtApiJack *) info->object; object->stopStream(); pthread_exit( NULL ); } bool RtApiJack :: callbackEvent( unsigned long nframes ) { if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; error( RtAudioError::WARNING ); return FAILURE; } if ( stream_.bufferSize != nframes ) { errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; error( RtAudioError::WARNING ); return FAILURE; } CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; JackHandle *handle = (JackHandle *) stream_.apiHandle; // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > 3 ) { ThreadHandle threadId; stream_.state = STREAM_STOPPING; if ( handle->internalDrain == true ) pthread_create( &threadId, NULL, jackStopStream, info ); else pthread_cond_signal( &handle->condition ); return SUCCESS; } // Invoke user callback first, to get fresh output data. if ( handle->drainCounter == 0 ) { RtAudioCallback callback = (RtAudioCallback) info->callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; if ( stream_.mode != INPUT && handle->xrun[0] == true ) { status |= RTAUDIO_OUTPUT_UNDERFLOW; handle->xrun[0] = false; } if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { status |= RTAUDIO_INPUT_OVERFLOW; handle->xrun[1] = false; } int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); if ( cbReturnValue == 2 ) { stream_.state = STREAM_STOPPING; handle->drainCounter = 2; ThreadHandle id; pthread_create( &id, NULL, jackStopStream, info ); return SUCCESS; } else if ( cbReturnValue == 1 ) { handle->drainCounter = 1; handle->internalDrain = true; } } jack_default_audio_sample_t *jackbuffer; unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( handle->drainCounter > 1 ) { // write zeros to the output stream for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); memset( jackbuffer, 0, bufferBytes ); } } else if ( stream_.doConvertBuffer[0] ) { convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); } } else { // no buffer conversion for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); } } } // Don't bother draining input if ( handle->drainCounter ) { handle->drainCounter++; goto unlock; } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { if ( stream_.doConvertBuffer[1] ) { for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); } convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); } else { // no buffer conversion for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); } } } unlock: RtApi::tickStreamTime(); return SUCCESS; } //******************** End of __UNIX_JACK__ *********************// #endif #if defined(__WINDOWS_ASIO__) // ASIO API on Windows // The ASIO API is designed around a callback scheme, so this // implementation is similar to that used for OS-X CoreAudio and Linux // Jack. The primary constraint with ASIO is that it only allows // access to a single driver at a time. Thus, it is not possible to // have more than one simultaneous RtAudio stream. // // This implementation also requires a number of external ASIO files // and a few global variables. The ASIO callback scheme does not // allow for the passing of user data, so we must create a global // pointer to our callbackInfo structure. // // On unix systems, we make use of a pthread condition variable. // Since there is no equivalent in Windows, I hacked something based // on information found in // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. #include "asiosys.h" #include "asio.h" #include "iasiothiscallresolver.h" #include "asiodrivers.h" #include static AsioDrivers drivers; static ASIOCallbacks asioCallbacks; static ASIODriverInfo driverInfo; static CallbackInfo *asioCallbackInfo; static bool asioXRun; struct AsioHandle { int drainCounter; // Tracks callback counts when draining bool internalDrain; // Indicates if stop is initiated from callback or not. ASIOBufferInfo *bufferInfos; HANDLE condition; AsioHandle() :drainCounter(0), internalDrain(false), bufferInfos(0) {} }; // Function declarations (definitions at end of section) static const char* getAsioErrorString( ASIOError result ); static void sampleRateChanged( ASIOSampleRate sRate ); static long asioMessages( long selector, long value, void* message, double* opt ); RtApiAsio :: RtApiAsio() { // ASIO cannot run on a multi-threaded appartment. You can call // CoInitialize beforehand, but it must be for appartment threading // (in which case, CoInitilialize will return S_FALSE here). coInitialized_ = false; HRESULT hr = CoInitialize( NULL ); if ( FAILED(hr) ) { errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; error( RtAudioError::WARNING ); } coInitialized_ = true; drivers.removeCurrentDriver(); driverInfo.asioVersion = 2; // See note in DirectSound implementation about GetDesktopWindow(). driverInfo.sysRef = GetForegroundWindow(); } RtApiAsio :: ~RtApiAsio() { if ( stream_.state != STREAM_CLOSED ) closeStream(); if ( coInitialized_ ) CoUninitialize(); } unsigned int RtApiAsio :: getDeviceCount( void ) { return (unsigned int) drivers.asioGetNumDev(); } RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; info.probed = false; // Get device ID unsigned int nDevices = getDeviceCount(); if ( nDevices == 0 ) { errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; error( RtAudioError::INVALID_USE ); return info; } if ( device >= nDevices ) { errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; error( RtAudioError::INVALID_USE ); return info; } // If a stream is already open, we cannot probe other devices. Thus, use the saved results. if ( stream_.state != STREAM_CLOSED ) { if ( device >= devices_.size() ) { errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; error( RtAudioError::WARNING ); return info; } return devices_[ device ]; } char driverName[32]; ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } info.name = driverName; if ( !drivers.loadDriver( driverName ) ) { errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } result = ASIOInit( &driverInfo ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Determine the device channel information. long inputChannels, outputChannels; result = ASIOGetChannels( &inputChannels, &outputChannels ); if ( result != ASE_OK ) { drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } info.outputChannels = outputChannels; info.inputChannels = inputChannels; if ( info.outputChannels > 0 && info.inputChannels > 0 ) info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; // Determine the supported sample rates. info.sampleRates.clear(); for ( unsigned int i=0; i info.preferredSampleRate ) ) info.preferredSampleRate = SAMPLE_RATES[i]; } } // Determine supported data types ... just check first channel and assume rest are the same. ASIOChannelInfo channelInfo; channelInfo.channel = 0; channelInfo.isInput = true; if ( info.inputChannels <= 0 ) channelInfo.isInput = false; result = ASIOGetChannelInfo( &channelInfo ); if ( result != ASE_OK ) { drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } info.nativeFormats = 0; if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) info.nativeFormats |= RTAUDIO_SINT16; else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) info.nativeFormats |= RTAUDIO_SINT32; else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) info.nativeFormats |= RTAUDIO_FLOAT32; else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) info.nativeFormats |= RTAUDIO_FLOAT64; else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) info.nativeFormats |= RTAUDIO_SINT24; if ( info.outputChannels > 0 ) if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; if ( info.inputChannels > 0 ) if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; info.probed = true; drivers.removeCurrentDriver(); return info; } static void bufferSwitch( long index, ASIOBool /*processNow*/ ) { RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; object->callbackEvent( index ); } void RtApiAsio :: saveDeviceInfo( void ) { devices_.clear(); unsigned int nDevices = getDeviceCount(); devices_.resize( nDevices ); for ( unsigned int i=0; isaveDeviceInfo(); if ( !drivers.loadDriver( driverName ) ) { errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; errorText_ = errorStream_.str(); return FAILURE; } result = ASIOInit( &driverInfo ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; errorText_ = errorStream_.str(); return FAILURE; } } // keep them before any "goto error", they are used for error cleanup + goto device boundary checks bool buffersAllocated = false; AsioHandle *handle = (AsioHandle *) stream_.apiHandle; unsigned int nChannels; // Check the device channel count. long inputChannels, outputChannels; result = ASIOGetChannels( &inputChannels, &outputChannels ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; errorText_ = errorStream_.str(); goto error; } if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; errorText_ = errorStream_.str(); goto error; } stream_.nDeviceChannels[mode] = channels; stream_.nUserChannels[mode] = channels; stream_.channelOffset[mode] = firstChannel; // Verify the sample rate is supported. result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; errorText_ = errorStream_.str(); goto error; } // Get the current sample rate ASIOSampleRate currentRate; result = ASIOGetSampleRate( ¤tRate ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; errorText_ = errorStream_.str(); goto error; } // Set the sample rate only if necessary if ( currentRate != sampleRate ) { result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; errorText_ = errorStream_.str(); goto error; } } // Determine the driver data type. ASIOChannelInfo channelInfo; channelInfo.channel = 0; if ( mode == OUTPUT ) channelInfo.isInput = false; else channelInfo.isInput = true; result = ASIOGetChannelInfo( &channelInfo ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; errorText_ = errorStream_.str(); goto error; } // Assuming WINDOWS host is always little-endian. stream_.doByteSwap[mode] = false; stream_.userFormat = format; stream_.deviceFormat[mode] = 0; if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { stream_.deviceFormat[mode] = RTAUDIO_SINT16; if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; } else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { stream_.deviceFormat[mode] = RTAUDIO_SINT32; if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; } else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; } else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; } else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) { stream_.deviceFormat[mode] = RTAUDIO_SINT24; if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true; } if ( stream_.deviceFormat[mode] == 0 ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; errorText_ = errorStream_.str(); goto error; } // Set the buffer size. For a duplex stream, this will end up // setting the buffer size based on the input constraints, which // should be ok. long minSize, maxSize, preferSize, granularity; result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; errorText_ = errorStream_.str(); goto error; } if ( isDuplexInput ) { // When this is the duplex input (output was opened before), then we have to use the same // buffersize as the output, because it might use the preferred buffer size, which most // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, // So instead of throwing an error, make them equal. The caller uses the reference // to the "bufferSize" param as usual to set up processing buffers. *bufferSize = stream_.bufferSize; } else { if ( *bufferSize == 0 ) *bufferSize = preferSize; else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; else if ( granularity == -1 ) { // Make sure bufferSize is a power of two. int log2_of_min_size = 0; int log2_of_max_size = 0; for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { if ( minSize & ((long)1 << i) ) log2_of_min_size = i; if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; } long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); int min_delta_num = log2_of_min_size; for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); if (current_delta < min_delta) { min_delta = current_delta; min_delta_num = i; } } *bufferSize = ( (unsigned int)1 << min_delta_num ); if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; } else if ( granularity != 0 ) { // Set to an even multiple of granularity, rounding up. *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; } } /* // we don't use it anymore, see above! // Just left it here for the case... if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; goto error; } */ stream_.bufferSize = *bufferSize; stream_.nBuffers = 2; if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; else stream_.userInterleaved = true; // ASIO always uses non-interleaved buffers. stream_.deviceInterleaved[mode] = false; // Allocate, if necessary, our AsioHandle structure for the stream. if ( handle == 0 ) { try { handle = new AsioHandle; } catch ( std::bad_alloc& ) { errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; goto error; } handle->bufferInfos = 0; // Create a manual-reset event. handle->condition = CreateEvent( NULL, // no security TRUE, // manual-reset FALSE, // non-signaled initially NULL ); // unnamed stream_.apiHandle = (void *) handle; } // Create the ASIO internal buffers. Since RtAudio sets up input // and output separately, we'll have to dispose of previously // created output buffers for a duplex stream. if ( mode == INPUT && stream_.mode == OUTPUT ) { ASIODisposeBuffers(); if ( handle->bufferInfos ) free( handle->bufferInfos ); } // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. unsigned int i; nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); if ( handle->bufferInfos == NULL ) { errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; errorText_ = errorStream_.str(); goto error; } ASIOBufferInfo *infos; infos = handle->bufferInfos; for ( i=0; iisInput = ASIOFalse; infos->channelNum = i + stream_.channelOffset[0]; infos->buffers[0] = infos->buffers[1] = 0; } for ( i=0; iisInput = ASIOTrue; infos->channelNum = i + stream_.channelOffset[1]; infos->buffers[0] = infos->buffers[1] = 0; } // prepare for callbacks stream_.sampleRate = sampleRate; stream_.device[mode] = device; stream_.mode = isDuplexInput ? DUPLEX : mode; // store this class instance before registering callbacks, that are going to use it asioCallbackInfo = &stream_.callbackInfo; stream_.callbackInfo.object = (void *) this; // Set up the ASIO callback structure and create the ASIO data buffers. asioCallbacks.bufferSwitch = &bufferSwitch; asioCallbacks.sampleRateDidChange = &sampleRateChanged; asioCallbacks.asioMessage = &asioMessages; asioCallbacks.bufferSwitchTimeInfo = NULL; result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); if ( result != ASE_OK ) { // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver // in that case, let's be naïve and try that instead *bufferSize = preferSize; stream_.bufferSize = *bufferSize; result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); } if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; errorText_ = errorStream_.str(); goto error; } buffersAllocated = true; stream_.state = STREAM_STOPPED; // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] ) stream_.doConvertBuffer[mode] = true; if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && stream_.nUserChannels[mode] > 1 ) stream_.doConvertBuffer[mode] = true; // Allocate necessary internal buffers unsigned long bufferBytes; bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; goto error; } if ( stream_.doConvertBuffer[mode] ) { bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); if ( isDuplexInput && stream_.deviceBuffer ) { unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); if ( bufferBytes <= bytesOut ) makeBuffer = false; } if ( makeBuffer ) { bufferBytes *= *bufferSize; if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); if ( stream_.deviceBuffer == NULL ) { errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; goto error; } } } // Determine device latencies long inputLatency, outputLatency; result = ASIOGetLatencies( &inputLatency, &outputLatency ); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING); // warn but don't fail } else { stream_.latency[0] = outputLatency; stream_.latency[1] = inputLatency; } // Setup the buffer conversion information structure. We don't use // buffers to do channel offsets, so we override that parameter // here. if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); return SUCCESS; error: if ( !isDuplexInput ) { // the cleanup for error in the duplex input, is done by RtApi::openStream // So we clean up for single channel only if ( buffersAllocated ) ASIODisposeBuffers(); drivers.removeCurrentDriver(); if ( handle ) { CloseHandle( handle->condition ); if ( handle->bufferInfos ) free( handle->bufferInfos ); delete handle; stream_.apiHandle = 0; } if ( stream_.userBuffer[mode] ) { free( stream_.userBuffer[mode] ); stream_.userBuffer[mode] = 0; } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } } return FAILURE; }//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// void RtApiAsio :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; error( RtAudioError::WARNING ); return; } if ( stream_.state == STREAM_RUNNING ) { stream_.state = STREAM_STOPPED; ASIOStop(); } ASIODisposeBuffers(); drivers.removeCurrentDriver(); AsioHandle *handle = (AsioHandle *) stream_.apiHandle; if ( handle ) { CloseHandle( handle->condition ); if ( handle->bufferInfos ) free( handle->bufferInfos ); delete handle; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } stream_.mode = UNINITIALIZED; stream_.state = STREAM_CLOSED; } bool stopThreadCalled = false; void RtApiAsio :: startStream() { verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAsio::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } AsioHandle *handle = (AsioHandle *) stream_.apiHandle; ASIOError result = ASIOStart(); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; errorText_ = errorStream_.str(); goto unlock; } handle->drainCounter = 0; handle->internalDrain = false; ResetEvent( handle->condition ); stream_.state = STREAM_RUNNING; asioXRun = false; unlock: stopThreadCalled = false; if ( result == ASE_OK ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiAsio :: stopStream() { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } AsioHandle *handle = (AsioHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( handle->drainCounter == 0 ) { handle->drainCounter = 2; WaitForSingleObject( handle->condition, INFINITE ); // block until signaled } } stream_.state = STREAM_STOPPED; ASIOError result = ASIOStop(); if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; errorText_ = errorStream_.str(); } if ( result == ASE_OK ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiAsio :: abortStream() { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } // The following lines were commented-out because some behavior was // noted where the device buffers need to be zeroed to avoid // continuing sound, even when the device buffers are completely // disposed. So now, calling abort is the same as calling stop. // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; // handle->drainCounter = 2; stopStream(); } // This function will be called by a spawned thread when the user // callback function signals that the stream should be stopped or // aborted. It is necessary to handle it this way because the // callbackEvent() function must return before the ASIOStop() // function will return. static unsigned __stdcall asioStopStream( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiAsio *object = (RtApiAsio *) info->object; object->stopStream(); _endthreadex( 0 ); return 0; } bool RtApiAsio :: callbackEvent( long bufferIndex ) { if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; error( RtAudioError::WARNING ); return FAILURE; } CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; AsioHandle *handle = (AsioHandle *) stream_.apiHandle; // Check if we were draining the stream and signal if finished. if ( handle->drainCounter > 3 ) { stream_.state = STREAM_STOPPING; if ( handle->internalDrain == false ) SetEvent( handle->condition ); else { // spawn a thread to stop the stream unsigned threadId; stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, &stream_.callbackInfo, 0, &threadId ); } return SUCCESS; } // Invoke user callback to get fresh output data UNLESS we are // draining stream. if ( handle->drainCounter == 0 ) { RtAudioCallback callback = (RtAudioCallback) info->callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; if ( stream_.mode != INPUT && asioXRun == true ) { status |= RTAUDIO_OUTPUT_UNDERFLOW; asioXRun = false; } if ( stream_.mode != OUTPUT && asioXRun == true ) { status |= RTAUDIO_INPUT_OVERFLOW; asioXRun = false; } int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); if ( cbReturnValue == 2 ) { stream_.state = STREAM_STOPPING; handle->drainCounter = 2; unsigned threadId; stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, &stream_.callbackInfo, 0, &threadId ); return SUCCESS; } else if ( cbReturnValue == 1 ) { handle->drainCounter = 1; handle->internalDrain = true; } } unsigned int nChannels, bufferBytes, i, j; nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); if ( handle->drainCounter > 1 ) { // write zeros to the output stream for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); } } else if ( stream_.doConvertBuffer[0] ) { convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); if ( stream_.doByteSwap[0] ) byteSwapBuffer( stream_.deviceBuffer, stream_.bufferSize * stream_.nDeviceChannels[0], stream_.deviceFormat[0] ); for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) memcpy( handle->bufferInfos[i].buffers[bufferIndex], &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); } } else { if ( stream_.doByteSwap[0] ) byteSwapBuffer( stream_.userBuffer[0], stream_.bufferSize * stream_.nUserChannels[0], stream_.userFormat ); for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) memcpy( handle->bufferInfos[i].buffers[bufferIndex], &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); } } } // Don't bother draining input if ( handle->drainCounter ) { handle->drainCounter++; goto unlock; } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); if (stream_.doConvertBuffer[1]) { // Always interleave ASIO input data. for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) memcpy( &stream_.deviceBuffer[j++*bufferBytes], handle->bufferInfos[i].buffers[bufferIndex], bufferBytes ); } if ( stream_.doByteSwap[1] ) byteSwapBuffer( stream_.deviceBuffer, stream_.bufferSize * stream_.nDeviceChannels[1], stream_.deviceFormat[1] ); convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); } else { for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { memcpy( &stream_.userBuffer[1][bufferBytes*j++], handle->bufferInfos[i].buffers[bufferIndex], bufferBytes ); } } if ( stream_.doByteSwap[1] ) byteSwapBuffer( stream_.userBuffer[1], stream_.bufferSize * stream_.nUserChannels[1], stream_.userFormat ); } } unlock: // The following call was suggested by Malte Clasen. While the API // documentation indicates it should not be required, some device // drivers apparently do not function correctly without it. ASIOOutputReady(); RtApi::tickStreamTime(); return SUCCESS; } static void sampleRateChanged( ASIOSampleRate sRate ) { // The ASIO documentation says that this usually only happens during // external sync. Audio processing is not stopped by the driver, // actual sample rate might not have even changed, maybe only the // sample rate status of an AES/EBU or S/PDIF digital input at the // audio device. RtApi *object = (RtApi *) asioCallbackInfo->object; try { object->stopStream(); } catch ( RtAudioError &exception ) { std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; return; } std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; } static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ ) { long ret = 0; switch( selector ) { case kAsioSelectorSupported: if ( value == kAsioResetRequest || value == kAsioEngineVersion || value == kAsioResyncRequest || value == kAsioLatenciesChanged // The following three were added for ASIO 2.0, you don't // necessarily have to support them. || value == kAsioSupportsTimeInfo || value == kAsioSupportsTimeCode || value == kAsioSupportsInputMonitor) ret = 1L; break; case kAsioResetRequest: // Defer the task and perform the reset of the driver during the // next "safe" situation. You cannot reset the driver right now, // as this code is called from the driver. Reset the driver is // done by completely destruct is. I.e. ASIOStop(), // ASIODisposeBuffers(), Destruction Afterwards you initialize the // driver again. std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; ret = 1L; break; case kAsioResyncRequest: // This informs the application that the driver encountered some // non-fatal data loss. It is used for synchronization purposes // of different media. Added mainly to work around the Win16Mutex // problems in Windows 95/98 with the Windows Multimedia system, // which could lose data because the Mutex was held too long by // another thread. However a driver can issue it in other // situations, too. // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; asioXRun = true; ret = 1L; break; case kAsioLatenciesChanged: // This will inform the host application that the drivers were // latencies changed. Beware, it this does not mean that the // buffer sizes have changed! You might need to update internal // delay data. std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; ret = 1L; break; case kAsioEngineVersion: // Return the supported ASIO version of the host application. If // a host application does not implement this selector, ASIO 1.0 // is assumed by the driver. ret = 2L; break; case kAsioSupportsTimeInfo: // Informs the driver whether the // asioCallbacks.bufferSwitchTimeInfo() callback is supported. // For compatibility with ASIO 1.0 drivers the host application // should always support the "old" bufferSwitch method, too. ret = 0; break; case kAsioSupportsTimeCode: // Informs the driver whether application is interested in time // code info. If an application does not need to know about time // code, the driver has less work to do. ret = 0; break; } return ret; } static const char* getAsioErrorString( ASIOError result ) { struct Messages { ASIOError value; const char*message; }; static const Messages m[] = { { ASE_NotPresent, "Hardware input or output is not present or available." }, { ASE_HWMalfunction, "Hardware is malfunctioning." }, { ASE_InvalidParameter, "Invalid input parameter." }, { ASE_InvalidMode, "Invalid mode." }, { ASE_SPNotAdvancing, "Sample position not advancing." }, { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, { ASE_NoMemory, "Not enough memory to complete the request." } }; for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) if ( m[i].value == result ) return m[i].message; return "Unknown error."; } //******************** End of __WINDOWS_ASIO__ *********************// #endif #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API // Authored by Marcus Tomlinson , April 2014 // - Introduces support for the Windows WASAPI API // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user #ifndef INITGUID #define INITGUID #endif #include #include #include #include //============================================================================= #define SAFE_RELEASE( objectPtr )\ if ( objectPtr )\ {\ objectPtr->Release();\ objectPtr = NULL;\ } typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex ); //----------------------------------------------------------------------------- // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size. // Therefore we must perform all necessary conversions to user buffers in order to satisfy these // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to // provide intermediate storage for read / write synchronization. class WasapiBuffer { public: WasapiBuffer() : buffer_( NULL ), bufferSize_( 0 ), inIndex_( 0 ), outIndex_( 0 ) {} ~WasapiBuffer() { free( buffer_ ); } // sets the length of the internal ring buffer void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { free( buffer_ ); buffer_ = ( char* ) calloc( bufferSize, formatBytes ); bufferSize_ = bufferSize; inIndex_ = 0; outIndex_ = 0; } // attempt to push a buffer into the ring buffer at the current "in" index bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) { if ( !buffer || // incoming buffer is NULL bufferSize == 0 || // incoming buffer has no data bufferSize > bufferSize_ ) // incoming buffer too large { return false; } unsigned int relOutIndex = outIndex_; unsigned int inIndexEnd = inIndex_ + bufferSize; if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) { relOutIndex += bufferSize_; } // "in" index can end on the "out" index but cannot begin at it if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) { return false; // not enough space between "in" index and "out" index } // copy buffer from external to internal int fromZeroSize = inIndex_ + bufferSize - bufferSize_; fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; int fromInSize = bufferSize - fromZeroSize; switch( format ) { case RTAUDIO_SINT8: memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) ); memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) ); break; case RTAUDIO_SINT16: memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) ); memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) ); break; case RTAUDIO_SINT24: memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) ); memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) ); break; case RTAUDIO_SINT32: memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) ); memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) ); break; case RTAUDIO_FLOAT32: memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) ); memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) ); break; case RTAUDIO_FLOAT64: memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) ); memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) ); break; } // update "in" index inIndex_ += bufferSize; inIndex_ %= bufferSize_; return true; } // attempt to pull a buffer from the ring buffer from the current "out" index bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) { if ( !buffer || // incoming buffer is NULL bufferSize == 0 || // incoming buffer has no data bufferSize > bufferSize_ ) // incoming buffer too large { return false; } unsigned int relInIndex = inIndex_; unsigned int outIndexEnd = outIndex_ + bufferSize; if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) { relInIndex += bufferSize_; } // "out" index can begin at and end on the "in" index if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) { return false; // not enough space between "out" index and "in" index } // copy buffer from internal to external int fromZeroSize = outIndex_ + bufferSize - bufferSize_; fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; int fromOutSize = bufferSize - fromZeroSize; switch( format ) { case RTAUDIO_SINT8: memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) ); memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) ); break; case RTAUDIO_SINT16: memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) ); memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) ); break; case RTAUDIO_SINT24: memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) ); memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) ); break; case RTAUDIO_SINT32: memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) ); memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) ); break; case RTAUDIO_FLOAT32: memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) ); memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) ); break; case RTAUDIO_FLOAT64: memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) ); memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) ); break; } // update "out" index outIndex_ += bufferSize; outIndex_ %= bufferSize_; return true; } private: char* buffer_; unsigned int bufferSize_; unsigned int inIndex_; unsigned int outIndex_; }; //----------------------------------------------------------------------------- // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate // between HW and the user. The convertBufferWasapi function is used to perform this conversion // between HwIn->UserIn and UserOut->HwOut during the stream callback loop. // This sample rate converter works best with conversions between one rate and its multiple. void convertBufferWasapi( char* outBuffer, const char* inBuffer, const unsigned int& channelCount, const unsigned int& inSampleRate, const unsigned int& outSampleRate, const unsigned int& inSampleCount, unsigned int& outSampleCount, const RtAudioFormat& format ) { // calculate the new outSampleCount and relative sampleStep float sampleRatio = ( float ) outSampleRate / inSampleRate; float sampleRatioInv = ( float ) 1 / sampleRatio; float sampleStep = 1.0f / sampleRatio; float inSampleFraction = 0.0f; outSampleCount = ( unsigned int ) std::roundf( inSampleCount * sampleRatio ); // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv ) { // frame-by-frame, copy each relative input sample into it's corresponding output sample for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ ) { unsigned int inSample = ( unsigned int ) inSampleFraction; switch ( format ) { case RTAUDIO_SINT8: memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) ); break; case RTAUDIO_SINT16: memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) ); break; case RTAUDIO_SINT24: memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) ); break; case RTAUDIO_SINT32: memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) ); break; case RTAUDIO_FLOAT32: memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) ); break; case RTAUDIO_FLOAT64: memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) ); break; } // jump to next in sample inSampleFraction += sampleStep; } } else // else interpolate { // frame-by-frame, copy each relative input sample into it's corresponding output sample for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ ) { unsigned int inSample = ( unsigned int ) inSampleFraction; float inSampleDec = inSampleFraction - inSample; unsigned int frameInSample = inSample * channelCount; unsigned int frameOutSample = outSample * channelCount; switch ( format ) { case RTAUDIO_SINT8: { for ( unsigned int channel = 0; channel < channelCount; channel++ ) { char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ]; char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ]; char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec ); ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; } break; } case RTAUDIO_SINT16: { for ( unsigned int channel = 0; channel < channelCount; channel++ ) { short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ]; short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ]; short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec ); ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; } break; } case RTAUDIO_SINT24: { for ( unsigned int channel = 0; channel < channelCount; channel++ ) { int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt(); int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt(); int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec ); ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; } break; } case RTAUDIO_SINT32: { for ( unsigned int channel = 0; channel < channelCount; channel++ ) { int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ]; int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ]; int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec ); ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; } break; } case RTAUDIO_FLOAT32: { for ( unsigned int channel = 0; channel < channelCount; channel++ ) { float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ]; float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ]; float sampleDiff = ( toSample - fromSample ) * inSampleDec; ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; } break; } case RTAUDIO_FLOAT64: { for ( unsigned int channel = 0; channel < channelCount; channel++ ) { double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ]; double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ]; double sampleDiff = ( toSample - fromSample ) * inSampleDec; ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; } break; } } // jump to next in sample inSampleFraction += sampleStep; } } } //----------------------------------------------------------------------------- // A structure to hold various information related to the WASAPI implementation. struct WasapiHandle { IAudioClient* captureAudioClient; IAudioClient* renderAudioClient; IAudioCaptureClient* captureClient; IAudioRenderClient* renderClient; HANDLE captureEvent; HANDLE renderEvent; WasapiHandle() : captureAudioClient( NULL ), renderAudioClient( NULL ), captureClient( NULL ), renderClient( NULL ), captureEvent( NULL ), renderEvent( NULL ) {} }; //============================================================================= RtApiWasapi::RtApiWasapi() : coInitialized_( false ), deviceEnumerator_( NULL ) { // WASAPI can run either apartment or multi-threaded HRESULT hr = CoInitialize( NULL ); if ( !FAILED( hr ) ) coInitialized_ = true; // Instantiate device enumerator hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL, CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ), ( void** ) &deviceEnumerator_ ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator"; error( RtAudioError::DRIVER_ERROR ); } } //----------------------------------------------------------------------------- RtApiWasapi::~RtApiWasapi() { if ( stream_.state != STREAM_CLOSED ) closeStream(); SAFE_RELEASE( deviceEnumerator_ ); // If this object previously called CoInitialize() if ( coInitialized_ ) CoUninitialize(); } //============================================================================= unsigned int RtApiWasapi::getDeviceCount( void ) { unsigned int captureDeviceCount = 0; unsigned int renderDeviceCount = 0; IMMDeviceCollection* captureDevices = NULL; IMMDeviceCollection* renderDevices = NULL; // Count capture devices errorText_.clear(); HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection."; goto Exit; } hr = captureDevices->GetCount( &captureDeviceCount ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count."; goto Exit; } // Count render devices hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection."; goto Exit; } hr = renderDevices->GetCount( &renderDeviceCount ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count."; goto Exit; } Exit: // release all references SAFE_RELEASE( captureDevices ); SAFE_RELEASE( renderDevices ); if ( errorText_.empty() ) return captureDeviceCount + renderDeviceCount; error( RtAudioError::DRIVER_ERROR ); return 0; } //----------------------------------------------------------------------------- RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; unsigned int captureDeviceCount = 0; unsigned int renderDeviceCount = 0; std::string defaultDeviceName; bool isCaptureDevice = false; PROPVARIANT deviceNameProp; PROPVARIANT defaultDeviceNameProp; IMMDeviceCollection* captureDevices = NULL; IMMDeviceCollection* renderDevices = NULL; IMMDevice* devicePtr = NULL; IMMDevice* defaultDevicePtr = NULL; IAudioClient* audioClient = NULL; IPropertyStore* devicePropStore = NULL; IPropertyStore* defaultDevicePropStore = NULL; WAVEFORMATEX* deviceFormat = NULL; WAVEFORMATEX* closestMatchFormat = NULL; // probed info.probed = false; // Count capture devices errorText_.clear(); RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection."; goto Exit; } hr = captureDevices->GetCount( &captureDeviceCount ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count."; goto Exit; } // Count render devices hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection."; goto Exit; } hr = renderDevices->GetCount( &renderDeviceCount ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; goto Exit; } // validate device index if ( device >= captureDeviceCount + renderDeviceCount ) { errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; errorType = RtAudioError::INVALID_USE; goto Exit; } // determine whether index falls within capture or render devices if ( device >= renderDeviceCount ) { hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle."; goto Exit; } isCaptureDevice = true; } else { hr = renderDevices->Item( device, &devicePtr ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle."; goto Exit; } isCaptureDevice = false; } // get default device name if ( isCaptureDevice ) { hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle."; goto Exit; } } else { hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle."; goto Exit; } } hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; goto Exit; } PropVariantInit( &defaultDeviceNameProp ); hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; goto Exit; } defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); // name hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; goto Exit; } PropVariantInit( &deviceNameProp ); hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; goto Exit; } info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); // is default if ( isCaptureDevice ) { info.isDefaultInput = info.name == defaultDeviceName; info.isDefaultOutput = false; } else { info.isDefaultInput = false; info.isDefaultOutput = info.name == defaultDeviceName; } // channel count hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client."; goto Exit; } hr = audioClient->GetMixFormat( &deviceFormat ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; goto Exit; } if ( isCaptureDevice ) { info.inputChannels = deviceFormat->nChannels; info.outputChannels = 0; info.duplexChannels = 0; } else { info.inputChannels = 0; info.outputChannels = deviceFormat->nChannels; info.duplexChannels = 0; } // sample rates info.sampleRates.clear(); // allow support for all sample rates as we have a built-in sample rate converter for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) { info.sampleRates.push_back( SAMPLE_RATES[i] ); } info.preferredSampleRate = deviceFormat->nSamplesPerSec; // native format info.nativeFormats = 0; if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT || ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) ) { if ( deviceFormat->wBitsPerSample == 32 ) { info.nativeFormats |= RTAUDIO_FLOAT32; } else if ( deviceFormat->wBitsPerSample == 64 ) { info.nativeFormats |= RTAUDIO_FLOAT64; } } else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM || ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) ) { if ( deviceFormat->wBitsPerSample == 8 ) { info.nativeFormats |= RTAUDIO_SINT8; } else if ( deviceFormat->wBitsPerSample == 16 ) { info.nativeFormats |= RTAUDIO_SINT16; } else if ( deviceFormat->wBitsPerSample == 24 ) { info.nativeFormats |= RTAUDIO_SINT24; } else if ( deviceFormat->wBitsPerSample == 32 ) { info.nativeFormats |= RTAUDIO_SINT32; } } // probed info.probed = true; Exit: // release all references PropVariantClear( &deviceNameProp ); PropVariantClear( &defaultDeviceNameProp ); SAFE_RELEASE( captureDevices ); SAFE_RELEASE( renderDevices ); SAFE_RELEASE( devicePtr ); SAFE_RELEASE( defaultDevicePtr ); SAFE_RELEASE( audioClient ); SAFE_RELEASE( devicePropStore ); SAFE_RELEASE( defaultDevicePropStore ); CoTaskMemFree( deviceFormat ); CoTaskMemFree( closestMatchFormat ); if ( !errorText_.empty() ) error( errorType ); return info; } //----------------------------------------------------------------------------- unsigned int RtApiWasapi::getDefaultOutputDevice( void ) { for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { if ( getDeviceInfo( i ).isDefaultOutput ) { return i; } } return 0; } //----------------------------------------------------------------------------- unsigned int RtApiWasapi::getDefaultInputDevice( void ) { for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { if ( getDeviceInfo( i ).isDefaultInput ) { return i; } } return 0; } //----------------------------------------------------------------------------- void RtApiWasapi::closeStream( void ) { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiWasapi::closeStream: No open stream to close."; error( RtAudioError::WARNING ); return; } if ( stream_.state != STREAM_STOPPED ) stopStream(); // clean up stream memory SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient ) SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient ) if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ) CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ); if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ) CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ); delete ( WasapiHandle* ) stream_.apiHandle; stream_.apiHandle = NULL; for ( int i = 0; i < 2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } // update stream state stream_.state = STREAM_CLOSED; } //----------------------------------------------------------------------------- void RtApiWasapi::startStream( void ) { verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiWasapi::startStream: The stream is already running."; error( RtAudioError::WARNING ); return; } // update stream state stream_.state = STREAM_RUNNING; // create WASAPI stream thread stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL ); if ( !stream_.callbackInfo.thread ) { errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread."; error( RtAudioError::THREAD_ERROR ); } else { SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority ); ResumeThread( ( void* ) stream_.callbackInfo.thread ); } } //----------------------------------------------------------------------------- void RtApiWasapi::stopStream( void ) { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiWasapi::stopStream: The stream is already stopped."; error( RtAudioError::WARNING ); return; } // inform stream thread by setting stream state to STREAM_STOPPING stream_.state = STREAM_STOPPING; // wait until stream thread is stopped while( stream_.state != STREAM_STOPPED ) { Sleep( 1 ); } // Wait for the last buffer to play before stopping. Sleep( 1000 * stream_.bufferSize / stream_.sampleRate ); // stop capture client if applicable if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream."; error( RtAudioError::DRIVER_ERROR ); return; } } // stop render client if applicable if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream."; error( RtAudioError::DRIVER_ERROR ); return; } } // close thread handle if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread."; error( RtAudioError::THREAD_ERROR ); return; } stream_.callbackInfo.thread = (ThreadHandle) NULL; } //----------------------------------------------------------------------------- void RtApiWasapi::abortStream( void ) { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiWasapi::abortStream: The stream is already stopped."; error( RtAudioError::WARNING ); return; } // inform stream thread by setting stream state to STREAM_STOPPING stream_.state = STREAM_STOPPING; // wait until stream thread is stopped while ( stream_.state != STREAM_STOPPED ) { Sleep( 1 ); } // stop capture client if applicable if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream."; error( RtAudioError::DRIVER_ERROR ); return; } } // stop render client if applicable if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream."; error( RtAudioError::DRIVER_ERROR ); return; } } // close thread handle if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; error( RtAudioError::THREAD_ERROR ); return; } stream_.callbackInfo.thread = (ThreadHandle) NULL; } //----------------------------------------------------------------------------- bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int* bufferSize, RtAudio::StreamOptions* options ) { bool methodResult = FAILURE; unsigned int captureDeviceCount = 0; unsigned int renderDeviceCount = 0; IMMDeviceCollection* captureDevices = NULL; IMMDeviceCollection* renderDevices = NULL; IMMDevice* devicePtr = NULL; WAVEFORMATEX* deviceFormat = NULL; unsigned int bufferBytes; stream_.state = STREAM_STOPPED; // create API Handle if not already created if ( !stream_.apiHandle ) stream_.apiHandle = ( void* ) new WasapiHandle(); // Count capture devices errorText_.clear(); RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection."; goto Exit; } hr = captureDevices->GetCount( &captureDeviceCount ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; goto Exit; } // Count render devices hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection."; goto Exit; } hr = renderDevices->GetCount( &renderDeviceCount ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; goto Exit; } // validate device index if ( device >= captureDeviceCount + renderDeviceCount ) { errorType = RtAudioError::INVALID_USE; errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; goto Exit; } // determine whether index falls within capture or render devices if ( device >= renderDeviceCount ) { if ( mode != INPUT ) { errorType = RtAudioError::INVALID_USE; errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device."; goto Exit; } // retrieve captureAudioClient from devicePtr IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; goto Exit; } hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &captureAudioClient ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; goto Exit; } hr = captureAudioClient->GetMixFormat( &deviceFormat ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; goto Exit; } stream_.nDeviceChannels[mode] = deviceFormat->nChannels; captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); } else { if ( mode != OUTPUT ) { errorType = RtAudioError::INVALID_USE; errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device."; goto Exit; } // retrieve renderAudioClient from devicePtr IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; hr = renderDevices->Item( device, &devicePtr ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; goto Exit; } hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &renderAudioClient ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; goto Exit; } hr = renderAudioClient->GetMixFormat( &deviceFormat ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; goto Exit; } stream_.nDeviceChannels[mode] = deviceFormat->nChannels; renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); } // fill stream data if ( ( stream_.mode == OUTPUT && mode == INPUT ) || ( stream_.mode == INPUT && mode == OUTPUT ) ) { stream_.mode = DUPLEX; } else { stream_.mode = mode; } stream_.device[mode] = device; stream_.doByteSwap[mode] = false; stream_.sampleRate = sampleRate; stream_.bufferSize = *bufferSize; stream_.nBuffers = 1; stream_.nUserChannels[mode] = channels; stream_.channelOffset[mode] = firstChannel; stream_.userFormat = format; stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats; if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; else stream_.userInterleaved = true; stream_.deviceInterleaved[mode] = true; // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] || stream_.nUserChannels != stream_.nDeviceChannels ) stream_.doConvertBuffer[mode] = true; else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && stream_.nUserChannels[mode] > 1 ) stream_.doConvertBuffer[mode] = true; if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); // Allocate necessary internal buffers bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 ); if ( !stream_.userBuffer[mode] ) { errorType = RtAudioError::MEMORY_ERROR; errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory."; goto Exit; } if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) stream_.callbackInfo.priority = 15; else stream_.callbackInfo.priority = 0; ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode methodResult = SUCCESS; Exit: //clean up SAFE_RELEASE( captureDevices ); SAFE_RELEASE( renderDevices ); SAFE_RELEASE( devicePtr ); CoTaskMemFree( deviceFormat ); // if method failed, close the stream if ( methodResult == FAILURE ) closeStream(); if ( !errorText_.empty() ) error( errorType ); return methodResult; } //============================================================================= DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr ) { if ( wasapiPtr ) ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread(); return 0; } DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr ) { if ( wasapiPtr ) ( ( RtApiWasapi* ) wasapiPtr )->stopStream(); return 0; } DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr ) { if ( wasapiPtr ) ( ( RtApiWasapi* ) wasapiPtr )->abortStream(); return 0; } //----------------------------------------------------------------------------- void RtApiWasapi::wasapiThread() { // as this is a new thread, we must CoInitialize it CoInitialize( NULL ); HRESULT hr; IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient; IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient; HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent; HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent; WAVEFORMATEX* captureFormat = NULL; WAVEFORMATEX* renderFormat = NULL; float captureSrRatio = 0.0f; float renderSrRatio = 0.0f; WasapiBuffer captureBuffer; WasapiBuffer renderBuffer; // declare local stream variables RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback; BYTE* streamBuffer = NULL; unsigned long captureFlags = 0; unsigned int bufferFrameCount = 0; unsigned int numFramesPadding = 0; unsigned int convBufferSize = 0; bool callbackPushed = false; bool callbackPulled = false; bool callbackStopped = false; int callbackResult = 0; // convBuffer is used to store converted buffers between WASAPI and the user char* convBuffer = NULL; unsigned int convBuffSize = 0; unsigned int deviceBuffSize = 0; errorText_.clear(); RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; // Attempt to assign "Pro Audio" characteristic to thread HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" ); if ( AvrtDll ) { DWORD taskIndex = 0; TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex ); FreeLibrary( AvrtDll ); } // start capture stream if applicable if ( captureAudioClient ) { hr = captureAudioClient->GetMixFormat( &captureFormat ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; goto Exit; } captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate ); // initialize capture stream according to desire buffer size float desiredBufferSize = stream_.bufferSize * captureSrRatio; REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec ); if ( !captureClient ) { hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK, desiredBufferPeriod, desiredBufferPeriod, captureFormat, NULL ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; goto Exit; } hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), ( void** ) &captureClient ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; goto Exit; } // configure captureEvent to trigger on every available capture buffer captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); if ( !captureEvent ) { errorType = RtAudioError::SYSTEM_ERROR; errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event."; goto Exit; } hr = captureAudioClient->SetEventHandle( captureEvent ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; goto Exit; } ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; } unsigned int inBufferSize = 0; hr = captureAudioClient->GetBufferSize( &inBufferSize ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; goto Exit; } // scale outBufferSize according to stream->user sample rate ratio unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT]; inBufferSize *= stream_.nDeviceChannels[INPUT]; // set captureBuffer size captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) ); // reset the capture stream hr = captureAudioClient->Reset(); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; goto Exit; } // start the capture stream hr = captureAudioClient->Start(); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream."; goto Exit; } } // start render stream if applicable if ( renderAudioClient ) { hr = renderAudioClient->GetMixFormat( &renderFormat ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; goto Exit; } renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); // initialize render stream according to desire buffer size float desiredBufferSize = stream_.bufferSize * renderSrRatio; REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec ); if ( !renderClient ) { hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK, desiredBufferPeriod, desiredBufferPeriod, renderFormat, NULL ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; goto Exit; } hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), ( void** ) &renderClient ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; goto Exit; } // configure renderEvent to trigger on every available render buffer renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); if ( !renderEvent ) { errorType = RtAudioError::SYSTEM_ERROR; errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event."; goto Exit; } hr = renderAudioClient->SetEventHandle( renderEvent ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle."; goto Exit; } ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; } unsigned int outBufferSize = 0; hr = renderAudioClient->GetBufferSize( &outBufferSize ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size."; goto Exit; } // scale inBufferSize according to user->stream sample rate ratio unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT]; outBufferSize *= stream_.nDeviceChannels[OUTPUT]; // set renderBuffer size renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) ); // reset the render stream hr = renderAudioClient->Reset(); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream."; goto Exit; } // start the render stream hr = renderAudioClient->Start(); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream."; goto Exit; } } if ( stream_.mode == INPUT ) { convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); } else if ( stream_.mode == OUTPUT ) { convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); } else if ( stream_.mode == DUPLEX ) { convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); } convBuffer = ( char* ) malloc( convBuffSize ); stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize ); if ( !convBuffer || !stream_.deviceBuffer ) { errorType = RtAudioError::MEMORY_ERROR; errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; goto Exit; } // stream process loop while ( stream_.state != STREAM_STOPPING ) { if ( !callbackPulled ) { // Callback Input // ============== // 1. Pull callback buffer from inputBuffer // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count // Convert callback buffer to user format if ( captureAudioClient ) { // Pull callback buffer from inputBuffer callbackPulled = captureBuffer.pullBuffer( convBuffer, ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT], stream_.deviceFormat[INPUT] ); if ( callbackPulled ) { // Convert callback buffer to user sample rate convertBufferWasapi( stream_.deviceBuffer, convBuffer, stream_.nDeviceChannels[INPUT], captureFormat->nSamplesPerSec, stream_.sampleRate, ( unsigned int ) ( stream_.bufferSize * captureSrRatio ), convBufferSize, stream_.deviceFormat[INPUT] ); if ( stream_.doConvertBuffer[INPUT] ) { // Convert callback buffer to user format convertBuffer( stream_.userBuffer[INPUT], stream_.deviceBuffer, stream_.convertInfo[INPUT] ); } else { // no further conversion, simple copy deviceBuffer to userBuffer memcpy( stream_.userBuffer[INPUT], stream_.deviceBuffer, stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) ); } } } else { // if there is no capture stream, set callbackPulled flag callbackPulled = true; } // Execute Callback // ================ // 1. Execute user callback method // 2. Handle return value from callback // if callback has not requested the stream to stop if ( callbackPulled && !callbackStopped ) { // Execute user callback method callbackResult = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT], stream_.bufferSize, getStreamTime(), captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0, stream_.callbackInfo.userData ); // Handle return value from callback if ( callbackResult == 1 ) { // instantiate a thread to stop this thread HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL ); if ( !threadHandle ) { errorType = RtAudioError::THREAD_ERROR; errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread."; goto Exit; } else if ( !CloseHandle( threadHandle ) ) { errorType = RtAudioError::THREAD_ERROR; errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; goto Exit; } callbackStopped = true; } else if ( callbackResult == 2 ) { // instantiate a thread to stop this thread HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); if ( !threadHandle ) { errorType = RtAudioError::THREAD_ERROR; errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; goto Exit; } else if ( !CloseHandle( threadHandle ) ) { errorType = RtAudioError::THREAD_ERROR; errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; goto Exit; } callbackStopped = true; } } } // Callback Output // =============== // 1. Convert callback buffer to stream format // 2. Convert callback buffer to stream sample rate and channel count // 3. Push callback buffer into outputBuffer if ( renderAudioClient && callbackPulled ) { if ( stream_.doConvertBuffer[OUTPUT] ) { // Convert callback buffer to stream format convertBuffer( stream_.deviceBuffer, stream_.userBuffer[OUTPUT], stream_.convertInfo[OUTPUT] ); } // Convert callback buffer to stream sample rate convertBufferWasapi( convBuffer, stream_.deviceBuffer, stream_.nDeviceChannels[OUTPUT], stream_.sampleRate, renderFormat->nSamplesPerSec, stream_.bufferSize, convBufferSize, stream_.deviceFormat[OUTPUT] ); // Push callback buffer into outputBuffer callbackPushed = renderBuffer.pushBuffer( convBuffer, convBufferSize * stream_.nDeviceChannels[OUTPUT], stream_.deviceFormat[OUTPUT] ); } else { // if there is no render stream, set callbackPushed flag callbackPushed = true; } // Stream Capture // ============== // 1. Get capture buffer from stream // 2. Push capture buffer into inputBuffer // 3. If 2. was successful: Release capture buffer if ( captureAudioClient ) { // if the callback input buffer was not pulled from captureBuffer, wait for next capture event if ( !callbackPulled ) { WaitForSingleObject( captureEvent, INFINITE ); } // Get capture buffer from stream hr = captureClient->GetBuffer( &streamBuffer, &bufferFrameCount, &captureFlags, NULL, NULL ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; goto Exit; } if ( bufferFrameCount != 0 ) { // Push capture buffer into inputBuffer if ( captureBuffer.pushBuffer( ( char* ) streamBuffer, bufferFrameCount * stream_.nDeviceChannels[INPUT], stream_.deviceFormat[INPUT] ) ) { // Release capture buffer hr = captureClient->ReleaseBuffer( bufferFrameCount ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; goto Exit; } } else { // Inform WASAPI that capture was unsuccessful hr = captureClient->ReleaseBuffer( 0 ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; goto Exit; } } } else { // Inform WASAPI that capture was unsuccessful hr = captureClient->ReleaseBuffer( 0 ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; goto Exit; } } } // Stream Render // ============= // 1. Get render buffer from stream // 2. Pull next buffer from outputBuffer // 3. If 2. was successful: Fill render buffer with next buffer // Release render buffer if ( renderAudioClient ) { // if the callback output buffer was not pushed to renderBuffer, wait for next render event if ( callbackPulled && !callbackPushed ) { WaitForSingleObject( renderEvent, INFINITE ); } // Get render buffer from stream hr = renderAudioClient->GetBufferSize( &bufferFrameCount ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; goto Exit; } hr = renderAudioClient->GetCurrentPadding( &numFramesPadding ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; goto Exit; } bufferFrameCount -= numFramesPadding; if ( bufferFrameCount != 0 ) { hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; goto Exit; } // Pull next buffer from outputBuffer // Fill render buffer with next buffer if ( renderBuffer.pullBuffer( ( char* ) streamBuffer, bufferFrameCount * stream_.nDeviceChannels[OUTPUT], stream_.deviceFormat[OUTPUT] ) ) { // Release render buffer hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; goto Exit; } } else { // Inform WASAPI that render was unsuccessful hr = renderClient->ReleaseBuffer( 0, 0 ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; goto Exit; } } } else { // Inform WASAPI that render was unsuccessful hr = renderClient->ReleaseBuffer( 0, 0 ); if ( FAILED( hr ) ) { errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; goto Exit; } } } // if the callback buffer was pushed renderBuffer reset callbackPulled flag if ( callbackPushed ) { callbackPulled = false; // tick stream time RtApi::tickStreamTime(); } } Exit: // clean up CoTaskMemFree( captureFormat ); CoTaskMemFree( renderFormat ); free ( convBuffer ); CoUninitialize(); // update stream state stream_.state = STREAM_STOPPED; if ( errorText_.empty() ) return; else error( errorType ); } //******************** End of __WINDOWS_WASAPI__ *********************// #endif #if defined(__WINDOWS_DS__) // Windows DirectSound API // Modified by Robin Davies, October 2005 // - Improvements to DirectX pointer chasing. // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. // - Auto-call CoInitialize for DSOUND and ASIO platforms. // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 // Changed device query structure for RtAudio 4.0.7, January 2010 #include #include #include #include #include #if defined(__MINGW32__) // missing from latest mingw winapi #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ #endif #define MINIMUM_DEVICE_BUFFER_SIZE 32768 #ifdef _MSC_VER // if Microsoft Visual C++ #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. #endif static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) { if ( pointer > bufferSize ) pointer -= bufferSize; if ( laterPointer < earlierPointer ) laterPointer += bufferSize; if ( pointer < earlierPointer ) pointer += bufferSize; return pointer >= earlierPointer && pointer < laterPointer; } // A structure to hold various information related to the DirectSound // API implementation. struct DsHandle { unsigned int drainCounter; // Tracks callback counts when draining bool internalDrain; // Indicates if stop is initiated from callback or not. void *id[2]; void *buffer[2]; bool xrun[2]; UINT bufferPointer[2]; DWORD dsBufferSize[2]; DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. HANDLE condition; DsHandle() :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } }; // Declarations for utility functions, callbacks, and structures // specific to the DirectSound implementation. static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, LPCTSTR description, LPCTSTR module, LPVOID lpContext ); static const char* getErrorString( int code ); static unsigned __stdcall callbackHandler( void *ptr ); struct DsDevice { LPGUID id[2]; bool validId[2]; bool found; std::string name; DsDevice() : found(false) { validId[0] = false; validId[1] = false; } }; struct DsProbeData { bool isInput; std::vector* dsDevices; }; RtApiDs :: RtApiDs() { // Dsound will run both-threaded. If CoInitialize fails, then just // accept whatever the mainline chose for a threading model. coInitialized_ = false; HRESULT hr = CoInitialize( NULL ); if ( !FAILED( hr ) ) coInitialized_ = true; } RtApiDs :: ~RtApiDs() { if ( stream_.state != STREAM_CLOSED ) closeStream(); if ( coInitialized_ ) CoUninitialize(); // balanced call. } // The DirectSound default output is always the first device. unsigned int RtApiDs :: getDefaultOutputDevice( void ) { return 0; } // The DirectSound default input is always the first input device, // which is the first capture device enumerated. unsigned int RtApiDs :: getDefaultInputDevice( void ) { return 0; } unsigned int RtApiDs :: getDeviceCount( void ) { // Set query flag for previously found devices to false, so that we // can check for any devices that have disappeared. for ( unsigned int i=0; i(dsDevices.size()); } RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; info.probed = false; if ( dsDevices.size() == 0 ) { // Force a query of all devices getDeviceCount(); if ( dsDevices.size() == 0 ) { errorText_ = "RtApiDs::getDeviceInfo: no devices found!"; error( RtAudioError::INVALID_USE ); return info; } } if ( device >= dsDevices.size() ) { errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; error( RtAudioError::INVALID_USE ); return info; } HRESULT result; if ( dsDevices[ device ].validId[0] == false ) goto probeInput; LPDIRECTSOUND output; DSCAPS outCaps; result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); goto probeInput; } outCaps.dwSize = sizeof( outCaps ); result = output->GetCaps( &outCaps ); if ( FAILED( result ) ) { output->Release(); errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); goto probeInput; } // Get output channel information. info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; // Get sample rate information. info.sampleRates.clear(); for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) { info.sampleRates.push_back( SAMPLE_RATES[k] ); if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) info.preferredSampleRate = SAMPLE_RATES[k]; } } // Get format information. if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; output->Release(); if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; if ( dsDevices[ device ].validId[1] == false ) { info.name = dsDevices[ device ].name; info.probed = true; return info; } probeInput: LPDIRECTSOUNDCAPTURE input; result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } DSCCAPS inCaps; inCaps.dwSize = sizeof( inCaps ); result = input->GetCaps( &inCaps ); if ( FAILED( result ) ) { input->Release(); errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Get input channel information. info.inputChannels = inCaps.dwChannels; // Get sample rate and format information. std::vector rates; if ( inCaps.dwChannels >= 2 ) { if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; if ( info.nativeFormats & RTAUDIO_SINT16 ) { if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 ); if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 ); if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 ); if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 ); } else if ( info.nativeFormats & RTAUDIO_SINT8 ) { if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 ); if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 ); if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 ); if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 ); } } else if ( inCaps.dwChannels == 1 ) { if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; if ( info.nativeFormats & RTAUDIO_SINT16 ) { if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 ); if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 ); if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 ); if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 ); } else if ( info.nativeFormats & RTAUDIO_SINT8 ) { if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 ); if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 ); if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 ); if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 ); } } else info.inputChannels = 0; // technically, this would be an error input->Release(); if ( info.inputChannels == 0 ) return info; // Copy the supported rates to the info structure but avoid duplication. bool found; for ( unsigned int i=0; i 0 && info.inputChannels > 0 ) info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; if ( device == 0 ) info.isDefaultInput = true; // Copy name and return. info.name = dsDevices[ device ].name; info.probed = true; return info; } bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ) { if ( channels + firstChannel > 2 ) { errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; return FAILURE; } size_t nDevices = dsDevices.size(); if ( nDevices == 0 ) { // This should not happen because a check is made before this function is called. errorText_ = "RtApiDs::probeDeviceOpen: no devices found!"; return FAILURE; } if ( device >= nDevices ) { // This should not happen because a check is made before this function is called. errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!"; return FAILURE; } if ( mode == OUTPUT ) { if ( dsDevices[ device ].validId[0] == false ) { errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; errorText_ = errorStream_.str(); return FAILURE; } } else { // mode == INPUT if ( dsDevices[ device ].validId[1] == false ) { errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; errorText_ = errorStream_.str(); return FAILURE; } } // According to a note in PortAudio, using GetDesktopWindow() // instead of GetForegroundWindow() is supposed to avoid problems // that occur when the application's window is not the foreground // window. Also, if the application window closes before the // DirectSound buffer, DirectSound can crash. In the past, I had // problems when using GetDesktopWindow() but it seems fine now // (January 2010). I'll leave it commented here. // HWND hWnd = GetForegroundWindow(); HWND hWnd = GetDesktopWindow(); // Check the numberOfBuffers parameter and limit the lowest value to // two. This is a judgement call and a value of two is probably too // low for capture, but it should work for playback. int nBuffers = 0; if ( options ) nBuffers = options->numberOfBuffers; if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; if ( nBuffers < 2 ) nBuffers = 3; // Check the lower range of the user-specified buffer size and set // (arbitrarily) to a lower bound of 32. if ( *bufferSize < 32 ) *bufferSize = 32; // Create the wave format structure. The data format setting will // be determined later. WAVEFORMATEX waveFormat; ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); waveFormat.wFormatTag = WAVE_FORMAT_PCM; waveFormat.nChannels = channels + firstChannel; waveFormat.nSamplesPerSec = (unsigned long) sampleRate; // Determine the device buffer size. By default, we'll use the value // defined above (32K), but we will grow it to make allowances for // very large software buffer sizes. DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE; DWORD dsPointerLeadTime = 0; void *ohandle = 0, *bhandle = 0; HRESULT result; if ( mode == OUTPUT ) { LPDIRECTSOUND output; result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } DSCAPS outCaps; outCaps.dwSize = sizeof( outCaps ); result = output->GetCaps( &outCaps ); if ( FAILED( result ) ) { output->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } // Check channel information. if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback."; errorText_ = errorStream_.str(); return FAILURE; } // Check format information. Use 16-bit format unless not // supported or user requests 8-bit. if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { waveFormat.wBitsPerSample = 16; stream_.deviceFormat[mode] = RTAUDIO_SINT16; } else { waveFormat.wBitsPerSample = 8; stream_.deviceFormat[mode] = RTAUDIO_SINT8; } stream_.userFormat = format; // Update wave format structure and buffer information. waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; // If the user wants an even bigger buffer, increase the device buffer size accordingly. while ( dsPointerLeadTime * 2U > dsBufferSize ) dsBufferSize *= 2; // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); if ( FAILED( result ) ) { output->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } // Even though we will write to the secondary buffer, we need to // access the primary buffer to set the correct output format // (since the default is 8-bit, 22 kHz!). Setup the DS primary // buffer description. DSBUFFERDESC bufferDescription; ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); bufferDescription.dwSize = sizeof( DSBUFFERDESC ); bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; // Obtain the primary buffer LPDIRECTSOUNDBUFFER buffer; result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); if ( FAILED( result ) ) { output->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } // Set the primary DS buffer sound format. result = buffer->SetFormat( &waveFormat ); if ( FAILED( result ) ) { output->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } // Setup the secondary DS buffer description. ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); bufferDescription.dwSize = sizeof( DSBUFFERDESC ); bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_LOCHARDWARE ); // Force hardware mixing bufferDescription.dwBufferBytes = dsBufferSize; bufferDescription.lpwfxFormat = &waveFormat; // Try to create the secondary DS buffer. If that doesn't work, // try to use software mixing. Otherwise, there's a problem. result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); if ( FAILED( result ) ) { bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | DSBCAPS_GLOBALFOCUS | DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_LOCSOFTWARE ); // Force software mixing result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); if ( FAILED( result ) ) { output->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } } // Get the buffer size ... might be different from what we specified. DSBCAPS dsbcaps; dsbcaps.dwSize = sizeof( DSBCAPS ); result = buffer->GetCaps( &dsbcaps ); if ( FAILED( result ) ) { output->Release(); buffer->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } dsBufferSize = dsbcaps.dwBufferBytes; // Lock the DS buffer LPVOID audioPtr; DWORD dataLen; result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); if ( FAILED( result ) ) { output->Release(); buffer->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } // Zero the DS buffer ZeroMemory( audioPtr, dataLen ); // Unlock the DS buffer result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); if ( FAILED( result ) ) { output->Release(); buffer->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } ohandle = (void *) output; bhandle = (void *) buffer; } if ( mode == INPUT ) { LPDIRECTSOUNDCAPTURE input; result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } DSCCAPS inCaps; inCaps.dwSize = sizeof( inCaps ); result = input->GetCaps( &inCaps ); if ( FAILED( result ) ) { input->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } // Check channel information. if ( inCaps.dwChannels < channels + firstChannel ) { errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; return FAILURE; } // Check format information. Use 16-bit format unless user // requests 8-bit. DWORD deviceFormats; if ( channels + firstChannel == 2 ) { deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { waveFormat.wBitsPerSample = 8; stream_.deviceFormat[mode] = RTAUDIO_SINT8; } else { // assume 16-bit is supported waveFormat.wBitsPerSample = 16; stream_.deviceFormat[mode] = RTAUDIO_SINT16; } } else { // channel == 1 deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { waveFormat.wBitsPerSample = 8; stream_.deviceFormat[mode] = RTAUDIO_SINT8; } else { // assume 16-bit is supported waveFormat.wBitsPerSample = 16; stream_.deviceFormat[mode] = RTAUDIO_SINT16; } } stream_.userFormat = format; // Update wave format structure and buffer information. waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; // If the user wants an even bigger buffer, increase the device buffer size accordingly. while ( dsPointerLeadTime * 2U > dsBufferSize ) dsBufferSize *= 2; // Setup the secondary DS buffer description. DSCBUFFERDESC bufferDescription; ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); bufferDescription.dwFlags = 0; bufferDescription.dwReserved = 0; bufferDescription.dwBufferBytes = dsBufferSize; bufferDescription.lpwfxFormat = &waveFormat; // Create the capture buffer. LPDIRECTSOUNDCAPTUREBUFFER buffer; result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); if ( FAILED( result ) ) { input->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } // Get the buffer size ... might be different from what we specified. DSCBCAPS dscbcaps; dscbcaps.dwSize = sizeof( DSCBCAPS ); result = buffer->GetCaps( &dscbcaps ); if ( FAILED( result ) ) { input->Release(); buffer->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } dsBufferSize = dscbcaps.dwBufferBytes; // NOTE: We could have a problem here if this is a duplex stream // and the play and capture hardware buffer sizes are different // (I'm actually not sure if that is a problem or not). // Currently, we are not verifying that. // Lock the capture buffer LPVOID audioPtr; DWORD dataLen; result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); if ( FAILED( result ) ) { input->Release(); buffer->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } // Zero the buffer ZeroMemory( audioPtr, dataLen ); // Unlock the buffer result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); if ( FAILED( result ) ) { input->Release(); buffer->Release(); errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } ohandle = (void *) input; bhandle = (void *) buffer; } // Set various stream parameters DsHandle *handle = 0; stream_.nDeviceChannels[mode] = channels + firstChannel; stream_.nUserChannels[mode] = channels; stream_.bufferSize = *bufferSize; stream_.channelOffset[mode] = firstChannel; stream_.deviceInterleaved[mode] = true; if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; else stream_.userInterleaved = true; // Set flag for buffer conversion stream_.doConvertBuffer[mode] = false; if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) stream_.doConvertBuffer[mode] = true; if (stream_.userFormat != stream_.deviceFormat[mode]) stream_.doConvertBuffer[mode] = true; if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && stream_.nUserChannels[mode] > 1 ) stream_.doConvertBuffer[mode] = true; // Allocate necessary internal buffers long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; goto error; } if ( stream_.doConvertBuffer[mode] ) { bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); if ( mode == INPUT ) { if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; } } if ( makeBuffer ) { bufferBytes *= *bufferSize; if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); if ( stream_.deviceBuffer == NULL ) { errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; goto error; } } } // Allocate our DsHandle structures for the stream. if ( stream_.apiHandle == 0 ) { try { handle = new DsHandle; } catch ( std::bad_alloc& ) { errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; goto error; } // Create a manual-reset event. handle->condition = CreateEvent( NULL, // no security TRUE, // manual-reset FALSE, // non-signaled initially NULL ); // unnamed stream_.apiHandle = (void *) handle; } else handle = (DsHandle *) stream_.apiHandle; handle->id[mode] = ohandle; handle->buffer[mode] = bhandle; handle->dsBufferSize[mode] = dsBufferSize; handle->dsPointerLeadTime[mode] = dsPointerLeadTime; stream_.device[mode] = device; stream_.state = STREAM_STOPPED; if ( stream_.mode == OUTPUT && mode == INPUT ) // We had already set up an output stream. stream_.mode = DUPLEX; else stream_.mode = mode; stream_.nBuffers = nBuffers; stream_.sampleRate = sampleRate; // Setup the buffer conversion information structure. if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); // Setup the callback thread. if ( stream_.callbackInfo.isRunning == false ) { unsigned threadId; stream_.callbackInfo.isRunning = true; stream_.callbackInfo.object = (void *) this; stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, &stream_.callbackInfo, 0, &threadId ); if ( stream_.callbackInfo.thread == 0 ) { errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; goto error; } // Boost DS thread priority SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); } return SUCCESS; error: if ( handle ) { if ( handle->buffer[0] ) { // the object pointer can be NULL and valid LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; if ( buffer ) buffer->Release(); object->Release(); } if ( handle->buffer[1] ) { LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; if ( buffer ) buffer->Release(); object->Release(); } CloseHandle( handle->condition ); delete handle; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } stream_.state = STREAM_CLOSED; return FAILURE; } void RtApiDs :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiDs::closeStream(): no open stream to close!"; error( RtAudioError::WARNING ); return; } // Stop the callback thread. stream_.callbackInfo.isRunning = false; WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); CloseHandle( (HANDLE) stream_.callbackInfo.thread ); DsHandle *handle = (DsHandle *) stream_.apiHandle; if ( handle ) { if ( handle->buffer[0] ) { // the object pointer can be NULL and valid LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; if ( buffer ) { buffer->Stop(); buffer->Release(); } object->Release(); } if ( handle->buffer[1] ) { LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; if ( buffer ) { buffer->Stop(); buffer->Release(); } object->Release(); } CloseHandle( handle->condition ); delete handle; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } stream_.mode = UNINITIALIZED; stream_.state = STREAM_CLOSED; } void RtApiDs :: startStream() { verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiDs::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } DsHandle *handle = (DsHandle *) stream_.apiHandle; // Increase scheduler frequency on lesser windows (a side-effect of // increasing timer accuracy). On greater windows (Win2K or later), // this is already in effect. timeBeginPeriod( 1 ); buffersRolling = false; duplexPrerollBytes = 0; if ( stream_.mode == DUPLEX ) { // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); } HRESULT result = 0; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; errorText_ = errorStream_.str(); goto unlock; } } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; result = buffer->Start( DSCBSTART_LOOPING ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; errorText_ = errorStream_.str(); goto unlock; } } handle->drainCounter = 0; handle->internalDrain = false; ResetEvent( handle->condition ); stream_.state = STREAM_RUNNING; unlock: if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); } void RtApiDs :: stopStream() { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } HRESULT result = 0; LPVOID audioPtr; DWORD dataLen; DsHandle *handle = (DsHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( handle->drainCounter == 0 ) { handle->drainCounter = 2; WaitForSingleObject( handle->condition, INFINITE ); // block until signaled } stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); // Stop the buffer and clear memory LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; result = buffer->Stop(); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; errorText_ = errorStream_.str(); goto unlock; } // Lock the buffer and clear it so that if we start to play again, // we won't have old data playing. result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; errorText_ = errorStream_.str(); goto unlock; } // Zero the DS buffer ZeroMemory( audioPtr, dataLen ); // Unlock the DS buffer result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; errorText_ = errorStream_.str(); goto unlock; } // If we start playing again, we must begin at beginning of buffer. handle->bufferPointer[0] = 0; } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; audioPtr = NULL; dataLen = 0; stream_.state = STREAM_STOPPED; if ( stream_.mode != DUPLEX ) MUTEX_LOCK( &stream_.mutex ); result = buffer->Stop(); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; errorText_ = errorStream_.str(); goto unlock; } // Lock the buffer and clear it so that if we start to play again, // we won't have old data playing. result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; errorText_ = errorStream_.str(); goto unlock; } // Zero the DS buffer ZeroMemory( audioPtr, dataLen ); // Unlock the DS buffer result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; errorText_ = errorStream_.str(); goto unlock; } // If we start recording again, we must begin at beginning of buffer. handle->bufferPointer[1] = 0; } unlock: timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. MUTEX_UNLOCK( &stream_.mutex ); if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); } void RtApiDs :: abortStream() { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } DsHandle *handle = (DsHandle *) stream_.apiHandle; handle->drainCounter = 2; stopStream(); } void RtApiDs :: callbackEvent() { if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { Sleep( 50 ); // sleep 50 milliseconds return; } if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; error( RtAudioError::WARNING ); return; } CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; DsHandle *handle = (DsHandle *) stream_.apiHandle; // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > stream_.nBuffers + 2 ) { stream_.state = STREAM_STOPPING; if ( handle->internalDrain == false ) SetEvent( handle->condition ); else stopStream(); return; } // Invoke user callback to get fresh output data UNLESS we are // draining stream. if ( handle->drainCounter == 0 ) { RtAudioCallback callback = (RtAudioCallback) info->callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; if ( stream_.mode != INPUT && handle->xrun[0] == true ) { status |= RTAUDIO_OUTPUT_UNDERFLOW; handle->xrun[0] = false; } if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { status |= RTAUDIO_INPUT_OVERFLOW; handle->xrun[1] = false; } int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); if ( cbReturnValue == 2 ) { stream_.state = STREAM_STOPPING; handle->drainCounter = 2; abortStream(); return; } else if ( cbReturnValue == 1 ) { handle->drainCounter = 1; handle->internalDrain = true; } } HRESULT result; DWORD currentWritePointer, safeWritePointer; DWORD currentReadPointer, safeReadPointer; UINT nextWritePointer; LPVOID buffer1 = NULL; LPVOID buffer2 = NULL; DWORD bufferSize1 = 0; DWORD bufferSize2 = 0; char *buffer; long bufferBytes; MUTEX_LOCK( &stream_.mutex ); if ( stream_.state == STREAM_STOPPED ) { MUTEX_UNLOCK( &stream_.mutex ); return; } if ( buffersRolling == false ) { if ( stream_.mode == DUPLEX ) { //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); // It takes a while for the devices to get rolling. As a result, // there's no guarantee that the capture and write device pointers // will move in lockstep. Wait here for both devices to start // rolling, and then set our buffer pointers accordingly. // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 // bytes later than the write buffer. // Stub: a serious risk of having a pre-emptive scheduling round // take place between the two GetCurrentPosition calls... but I'm // really not sure how to solve the problem. Temporarily boost to // Realtime priority, maybe; but I'm not sure what priority the // DirectSound service threads run at. We *should* be roughly // within a ms or so of correct. LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; DWORD startSafeWritePointer, startSafeReadPointer; result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } while ( true ) { result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; Sleep( 1 ); } //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; handle->bufferPointer[1] = safeReadPointer; } else if ( stream_.mode == OUTPUT ) { // Set the proper nextWritePosition after initial startup. LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; } buffersRolling = true; } if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; if ( handle->drainCounter > 1 ) { // write zeros to the output stream bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; bufferBytes *= formatBytes( stream_.userFormat ); memset( stream_.userBuffer[0], 0, bufferBytes ); } // Setup parameters and do buffer conversion if necessary. if ( stream_.doConvertBuffer[0] ) { buffer = stream_.deviceBuffer; convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; bufferBytes *= formatBytes( stream_.deviceFormat[0] ); } else { buffer = stream_.userBuffer[0]; bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; bufferBytes *= formatBytes( stream_.userFormat ); } // No byte swapping necessary in DirectSound implementation. // Ahhh ... windoze. 16-bit data is signed but 8-bit data is // unsigned. So, we need to convert our signed 8-bit data here to // unsigned. if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) for ( int i=0; idsBufferSize[0]; nextWritePointer = handle->bufferPointer[0]; DWORD endWrite, leadPointer; while ( true ) { // Find out where the read and "safe write" pointers are. result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } // We will copy our output buffer into the region between // safeWritePointer and leadPointer. If leadPointer is not // beyond the next endWrite position, wait until it is. leadPointer = safeWritePointer + handle->dsPointerLeadTime[0]; //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl; if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize; if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset endWrite = nextWritePointer + bufferBytes; // Check whether the entire write region is behind the play pointer. if ( leadPointer >= endWrite ) break; // If we are here, then we must wait until the leadPointer advances // beyond the end of our next write region. We use the // Sleep() function to suspend operation until that happens. double millis = ( endWrite - leadPointer ) * 1000.0; millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); if ( millis < 1.0 ) millis = 1.0; Sleep( (DWORD) millis ); } if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { // We've strayed into the forbidden zone ... resync the read pointer. handle->xrun[0] = true; nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize; handle->bufferPointer[0] = nextWritePointer; endWrite = nextWritePointer + bufferBytes; } // Lock free space in the buffer result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1, &bufferSize1, &buffer2, &bufferSize2, 0 ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } // Copy our buffer into the DS buffer CopyMemory( buffer1, buffer, bufferSize1 ); if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); // Update our buffer offset and unlock sound buffer dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; handle->bufferPointer[0] = nextWritePointer; } // Don't bother draining input if ( handle->drainCounter ) { handle->drainCounter++; goto unlock; } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { // Setup parameters. if ( stream_.doConvertBuffer[1] ) { buffer = stream_.deviceBuffer; bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; bufferBytes *= formatBytes( stream_.deviceFormat[1] ); } else { buffer = stream_.userBuffer[1]; bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; bufferBytes *= formatBytes( stream_.userFormat ); } LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; long nextReadPointer = handle->bufferPointer[1]; DWORD dsBufferSize = handle->dsBufferSize[1]; // Find out where the write and "safe read" pointers are. result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset DWORD endRead = nextReadPointer + bufferBytes; // Handling depends on whether we are INPUT or DUPLEX. // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, // then a wait here will drag the write pointers into the forbidden zone. // // In DUPLEX mode, rather than wait, we will back off the read pointer until // it's in a safe position. This causes dropouts, but it seems to be the only // practical way to sync up the read and write pointers reliably, given the // the very complex relationship between phase and increment of the read and write // pointers. // // In order to minimize audible dropouts in DUPLEX mode, we will // provide a pre-roll period of 0.5 seconds in which we return // zeros from the read buffer while the pointers sync up. if ( stream_.mode == DUPLEX ) { if ( safeReadPointer < endRead ) { if ( duplexPrerollBytes <= 0 ) { // Pre-roll time over. Be more agressive. int adjustment = endRead-safeReadPointer; handle->xrun[1] = true; // Two cases: // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, // and perform fine adjustments later. // - small adjustments: back off by twice as much. if ( adjustment >= 2*bufferBytes ) nextReadPointer = safeReadPointer-2*bufferBytes; else nextReadPointer = safeReadPointer-bufferBytes-adjustment; if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; } else { // In pre=roll time. Just do it. nextReadPointer = safeReadPointer - bufferBytes; while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; } endRead = nextReadPointer + bufferBytes; } } else { // mode == INPUT while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { // See comments for playback. double millis = (endRead - safeReadPointer) * 1000.0; millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); if ( millis < 1.0 ) millis = 1.0; Sleep( (DWORD) millis ); // Wake up and find out where we are now. result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset } } // Lock free space in the buffer result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1, &bufferSize1, &buffer2, &bufferSize2, 0 ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } if ( duplexPrerollBytes <= 0 ) { // Copy our buffer into the DS buffer CopyMemory( buffer, buffer1, bufferSize1 ); if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); } else { memset( buffer, 0, bufferSize1 ); if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); duplexPrerollBytes -= bufferSize1 + bufferSize2; } // Update our buffer offset and unlock sound buffer nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize; dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } handle->bufferPointer[1] = nextReadPointer; // No byte swapping necessary in DirectSound implementation. // If necessary, convert 8-bit data from unsigned to signed. if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) for ( int j=0; jobject; bool* isRunning = &info->isRunning; while ( *isRunning == true ) { object->callbackEvent(); } _endthreadex( 0 ); return 0; } static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, LPCTSTR description, LPCTSTR /*module*/, LPVOID lpContext ) { struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; std::vector& dsDevices = *probeInfo.dsDevices; HRESULT hr; bool validDevice = false; if ( probeInfo.isInput == true ) { DSCCAPS caps; LPDIRECTSOUNDCAPTURE object; hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); if ( hr != DS_OK ) return TRUE; caps.dwSize = sizeof(caps); hr = object->GetCaps( &caps ); if ( hr == DS_OK ) { if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) validDevice = true; } object->Release(); } else { DSCAPS caps; LPDIRECTSOUND object; hr = DirectSoundCreate( lpguid, &object, NULL ); if ( hr != DS_OK ) return TRUE; caps.dwSize = sizeof(caps); hr = object->GetCaps( &caps ); if ( hr == DS_OK ) { if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) validDevice = true; } object->Release(); } // If good device, then save its name and guid. std::string name = convertCharPointerToStdString( description ); //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) if ( lpguid == NULL ) name = "Default Device"; if ( validDevice ) { for ( unsigned int i=0; i #include // A structure to hold various information related to the ALSA API // implementation. struct AlsaHandle { snd_pcm_t *handles[2]; bool synchronized; bool xrun[2]; pthread_cond_t runnable_cv; bool runnable; AlsaHandle() :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } }; static void *alsaCallbackHandler( void * ptr ); RtApiAlsa :: RtApiAlsa() { // Nothing to do here. } RtApiAlsa :: ~RtApiAlsa() { if ( stream_.state != STREAM_CLOSED ) closeStream(); } unsigned int RtApiAlsa :: getDeviceCount( void ) { unsigned nDevices = 0; int result, subdevice, card; char name[64]; snd_ctl_t *handle; // Count cards and devices card = -1; snd_card_next( &card ); while ( card >= 0 ) { sprintf( name, "hw:%d", card ); result = snd_ctl_open( &handle, name, 0 ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); goto nextcard; } subdevice = -1; while( 1 ) { result = snd_ctl_pcm_next_device( handle, &subdevice ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); break; } if ( subdevice < 0 ) break; nDevices++; } nextcard: snd_ctl_close( handle ); snd_card_next( &card ); } result = snd_ctl_open( &handle, "default", 0 ); if (result == 0) { nDevices++; snd_ctl_close( handle ); } return nDevices; } RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; info.probed = false; unsigned nDevices = 0; int result, subdevice, card; char name[64]; snd_ctl_t *chandle; // Count cards and devices card = -1; subdevice = -1; snd_card_next( &card ); while ( card >= 0 ) { sprintf( name, "hw:%d", card ); result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); goto nextcard; } subdevice = -1; while( 1 ) { result = snd_ctl_pcm_next_device( chandle, &subdevice ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); break; } if ( subdevice < 0 ) break; if ( nDevices == device ) { sprintf( name, "hw:%d,%d", card, subdevice ); goto foundDevice; } nDevices++; } nextcard: snd_ctl_close( chandle ); snd_card_next( &card ); } result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); if ( result == 0 ) { if ( nDevices == device ) { strcpy( name, "default" ); goto foundDevice; } nDevices++; } if ( nDevices == 0 ) { errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; error( RtAudioError::INVALID_USE ); return info; } if ( device >= nDevices ) { errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; error( RtAudioError::INVALID_USE ); return info; } foundDevice: // If a stream is already open, we cannot probe the stream devices. // Thus, use the saved results. if ( stream_.state != STREAM_CLOSED && ( stream_.device[0] == device || stream_.device[1] == device ) ) { snd_ctl_close( chandle ); if ( device >= devices_.size() ) { errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; error( RtAudioError::WARNING ); return info; } return devices_[ device ]; } int openMode = SND_PCM_ASYNC; snd_pcm_stream_t stream; snd_pcm_info_t *pcminfo; snd_pcm_info_alloca( &pcminfo ); snd_pcm_t *phandle; snd_pcm_hw_params_t *params; snd_pcm_hw_params_alloca( ¶ms ); // First try for playback unless default device (which has subdev -1) stream = SND_PCM_STREAM_PLAYBACK; snd_pcm_info_set_stream( pcminfo, stream ); if ( subdevice != -1 ) { snd_pcm_info_set_device( pcminfo, subdevice ); snd_pcm_info_set_subdevice( pcminfo, 0 ); result = snd_ctl_pcm_info( chandle, pcminfo ); if ( result < 0 ) { // Device probably doesn't support playback. goto captureProbe; } } result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); goto captureProbe; } // The device is open ... fill the parameter structure. result = snd_pcm_hw_params_any( phandle, params ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); goto captureProbe; } // Get output channel information. unsigned int value; result = snd_pcm_hw_params_get_channels_max( params, &value ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); goto captureProbe; } info.outputChannels = value; snd_pcm_close( phandle ); captureProbe: stream = SND_PCM_STREAM_CAPTURE; snd_pcm_info_set_stream( pcminfo, stream ); // Now try for capture unless default device (with subdev = -1) if ( subdevice != -1 ) { result = snd_ctl_pcm_info( chandle, pcminfo ); snd_ctl_close( chandle ); if ( result < 0 ) { // Device probably doesn't support capture. if ( info.outputChannels == 0 ) return info; goto probeParameters; } } else snd_ctl_close( chandle ); result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); if ( info.outputChannels == 0 ) return info; goto probeParameters; } // The device is open ... fill the parameter structure. result = snd_pcm_hw_params_any( phandle, params ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); if ( info.outputChannels == 0 ) return info; goto probeParameters; } result = snd_pcm_hw_params_get_channels_max( params, &value ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); if ( info.outputChannels == 0 ) return info; goto probeParameters; } info.inputChannels = value; snd_pcm_close( phandle ); // If device opens for both playback and capture, we determine the channels. if ( info.outputChannels > 0 && info.inputChannels > 0 ) info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; // ALSA doesn't provide default devices so we'll use the first available one. if ( device == 0 && info.outputChannels > 0 ) info.isDefaultOutput = true; if ( device == 0 && info.inputChannels > 0 ) info.isDefaultInput = true; probeParameters: // At this point, we just need to figure out the supported data // formats and sample rates. We'll proceed by opening the device in // the direction with the maximum number of channels, or playback if // they are equal. This might limit our sample rate options, but so // be it. if ( info.outputChannels >= info.inputChannels ) stream = SND_PCM_STREAM_PLAYBACK; else stream = SND_PCM_STREAM_CAPTURE; snd_pcm_info_set_stream( pcminfo, stream ); result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // The device is open ... fill the parameter structure. result = snd_pcm_hw_params_any( phandle, params ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Test our discrete set of sample rate values. info.sampleRates.clear(); for ( unsigned int i=0; i info.preferredSampleRate ) ) info.preferredSampleRate = SAMPLE_RATES[i]; } } if ( info.sampleRates.size() == 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Probe the supported data formats ... we don't care about endian-ness just yet snd_pcm_format_t format; info.nativeFormats = 0; format = SND_PCM_FORMAT_S8; if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) info.nativeFormats |= RTAUDIO_SINT8; format = SND_PCM_FORMAT_S16; if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) info.nativeFormats |= RTAUDIO_SINT16; format = SND_PCM_FORMAT_S24; if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) info.nativeFormats |= RTAUDIO_SINT24; format = SND_PCM_FORMAT_S32; if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) info.nativeFormats |= RTAUDIO_SINT32; format = SND_PCM_FORMAT_FLOAT; if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) info.nativeFormats |= RTAUDIO_FLOAT32; format = SND_PCM_FORMAT_FLOAT64; if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) info.nativeFormats |= RTAUDIO_FLOAT64; // Check that we have at least one supported format if ( info.nativeFormats == 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Get the device name char *cardname; result = snd_card_get_name( card, &cardname ); if ( result >= 0 ) { sprintf( name, "hw:%s,%d", cardname, subdevice ); free( cardname ); } info.name = name; // That's all ... close the device and return snd_pcm_close( phandle ); info.probed = true; return info; } void RtApiAlsa :: saveDeviceInfo( void ) { devices_.clear(); unsigned int nDevices = getDeviceCount(); devices_.resize( nDevices ); for ( unsigned int i=0; iflags & RTAUDIO_ALSA_USE_DEFAULT ) snprintf(name, sizeof(name), "%s", "default"); else { // Count cards and devices card = -1; snd_card_next( &card ); while ( card >= 0 ) { sprintf( name, "hw:%d", card ); result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } subdevice = -1; while( 1 ) { result = snd_ctl_pcm_next_device( chandle, &subdevice ); if ( result < 0 ) break; if ( subdevice < 0 ) break; if ( nDevices == device ) { sprintf( name, "hw:%d,%d", card, subdevice ); snd_ctl_close( chandle ); goto foundDevice; } nDevices++; } snd_ctl_close( chandle ); snd_card_next( &card ); } result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); if ( result == 0 ) { if ( nDevices == device ) { strcpy( name, "default" ); goto foundDevice; } nDevices++; } if ( nDevices == 0 ) { // This should not happen because a check is made before this function is called. errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; return FAILURE; } if ( device >= nDevices ) { // This should not happen because a check is made before this function is called. errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; return FAILURE; } } foundDevice: // The getDeviceInfo() function will not work for a device that is // already open. Thus, we'll probe the system before opening a // stream and save the results for use by getDeviceInfo(). if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once this->saveDeviceInfo(); snd_pcm_stream_t stream; if ( mode == OUTPUT ) stream = SND_PCM_STREAM_PLAYBACK; else stream = SND_PCM_STREAM_CAPTURE; snd_pcm_t *phandle; int openMode = SND_PCM_ASYNC; result = snd_pcm_open( &phandle, name, stream, openMode ); if ( result < 0 ) { if ( mode == OUTPUT ) errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; else errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; errorText_ = errorStream_.str(); return FAILURE; } // Fill the parameter structure. snd_pcm_hw_params_t *hw_params; snd_pcm_hw_params_alloca( &hw_params ); result = snd_pcm_hw_params_any( phandle, hw_params ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } #if defined(__RTAUDIO_DEBUG__) fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); snd_pcm_hw_params_dump( hw_params, out ); #endif // Set access ... check user preference. if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { stream_.userInterleaved = false; result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); if ( result < 0 ) { result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); stream_.deviceInterleaved[mode] = true; } else stream_.deviceInterleaved[mode] = false; } else { stream_.userInterleaved = true; result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); if ( result < 0 ) { result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); stream_.deviceInterleaved[mode] = false; } else stream_.deviceInterleaved[mode] = true; } if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } // Determine how to set the device format. stream_.userFormat = format; snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; if ( format == RTAUDIO_SINT8 ) deviceFormat = SND_PCM_FORMAT_S8; else if ( format == RTAUDIO_SINT16 ) deviceFormat = SND_PCM_FORMAT_S16; else if ( format == RTAUDIO_SINT24 ) deviceFormat = SND_PCM_FORMAT_S24; else if ( format == RTAUDIO_SINT32 ) deviceFormat = SND_PCM_FORMAT_S32; else if ( format == RTAUDIO_FLOAT32 ) deviceFormat = SND_PCM_FORMAT_FLOAT; else if ( format == RTAUDIO_FLOAT64 ) deviceFormat = SND_PCM_FORMAT_FLOAT64; if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { stream_.deviceFormat[mode] = format; goto setFormat; } // The user requested format is not natively supported by the device. deviceFormat = SND_PCM_FORMAT_FLOAT64; if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; goto setFormat; } deviceFormat = SND_PCM_FORMAT_FLOAT; if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; goto setFormat; } deviceFormat = SND_PCM_FORMAT_S32; if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { stream_.deviceFormat[mode] = RTAUDIO_SINT32; goto setFormat; } deviceFormat = SND_PCM_FORMAT_S24; if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { stream_.deviceFormat[mode] = RTAUDIO_SINT24; goto setFormat; } deviceFormat = SND_PCM_FORMAT_S16; if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { stream_.deviceFormat[mode] = RTAUDIO_SINT16; goto setFormat; } deviceFormat = SND_PCM_FORMAT_S8; if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { stream_.deviceFormat[mode] = RTAUDIO_SINT8; goto setFormat; } // If we get here, no supported format was found. snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; errorText_ = errorStream_.str(); return FAILURE; setFormat: result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } // Determine whether byte-swaping is necessary. stream_.doByteSwap[mode] = false; if ( deviceFormat != SND_PCM_FORMAT_S8 ) { result = snd_pcm_format_cpu_endian( deviceFormat ); if ( result == 0 ) stream_.doByteSwap[mode] = true; else if (result < 0) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } } // Set the sample rate. result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } // Determine the number of channels for this device. We support a possible // minimum device channel number > than the value requested by the user. stream_.nUserChannels[mode] = channels; unsigned int value; result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); unsigned int deviceChannels = value; if ( result < 0 || deviceChannels < channels + firstChannel ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } deviceChannels = value; if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; stream_.nDeviceChannels[mode] = deviceChannels; // Set the device channels. result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } // Set the buffer (or period) size. int dir = 0; snd_pcm_uframes_t periodSize = *bufferSize; result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } *bufferSize = periodSize; // Set the buffer number, which in ALSA is referred to as the "period". unsigned int periods = 0; if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers; if ( periods < 2 ) periods = 4; // a fairly safe default value result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } // If attempting to setup a duplex stream, the bufferSize parameter // MUST be the same in both directions! if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; errorText_ = errorStream_.str(); return FAILURE; } stream_.bufferSize = *bufferSize; // Install the hardware configuration result = snd_pcm_hw_params( phandle, hw_params ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } #if defined(__RTAUDIO_DEBUG__) fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); snd_pcm_hw_params_dump( hw_params, out ); #endif // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. snd_pcm_sw_params_t *sw_params = NULL; snd_pcm_sw_params_alloca( &sw_params ); snd_pcm_sw_params_current( phandle, sw_params ); snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); // The following two settings were suggested by Theo Veenker //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); // here are two options for a fix //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); snd_pcm_uframes_t val; snd_pcm_sw_params_get_boundary( sw_params, &val ); snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); result = snd_pcm_sw_params( phandle, sw_params ); if ( result < 0 ) { snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } #if defined(__RTAUDIO_DEBUG__) fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); snd_pcm_sw_params_dump( sw_params, out ); #endif // Set flags for buffer conversion stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] ) stream_.doConvertBuffer[mode] = true; if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) stream_.doConvertBuffer[mode] = true; if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && stream_.nUserChannels[mode] > 1 ) stream_.doConvertBuffer[mode] = true; // Allocate the ApiHandle if necessary and then save. AlsaHandle *apiInfo = 0; if ( stream_.apiHandle == 0 ) { try { apiInfo = (AlsaHandle *) new AlsaHandle; } catch ( std::bad_alloc& ) { errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; goto error; } if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) { errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; goto error; } stream_.apiHandle = (void *) apiInfo; apiInfo->handles[0] = 0; apiInfo->handles[1] = 0; } else { apiInfo = (AlsaHandle *) stream_.apiHandle; } apiInfo->handles[mode] = phandle; phandle = 0; // Allocate necessary internal buffers. unsigned long bufferBytes; bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; goto error; } if ( stream_.doConvertBuffer[mode] ) { bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); if ( mode == INPUT ) { if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); if ( bufferBytes <= bytesOut ) makeBuffer = false; } } if ( makeBuffer ) { bufferBytes *= *bufferSize; if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); if ( stream_.deviceBuffer == NULL ) { errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; goto error; } } } stream_.sampleRate = sampleRate; stream_.nBuffers = periods; stream_.device[mode] = device; stream_.state = STREAM_STOPPED; // Setup the buffer conversion information structure. if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); // Setup thread if necessary. if ( stream_.mode == OUTPUT && mode == INPUT ) { // We had already set up an output stream. stream_.mode = DUPLEX; // Link the streams if possible. apiInfo->synchronized = false; if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) apiInfo->synchronized = true; else { errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; error( RtAudioError::WARNING ); } } else { stream_.mode = mode; // Setup callback thread. stream_.callbackInfo.object = (void *) this; // Set the thread attributes for joinable and realtime scheduling // priority (optional). The higher priority will only take affect // if the program is run as root or suid. Note, under Linux // processes with CAP_SYS_NICE privilege, a user can change // scheduling policy and priority (thus need not be root). See // POSIX "capabilities". pthread_attr_t attr; pthread_attr_init( &attr ); pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { // We previously attempted to increase the audio callback priority // to SCHED_RR here via the attributes. However, while no errors // were reported in doing so, it did not work. So, now this is // done in the alsaCallbackHandler function. stream_.callbackInfo.doRealtime = true; int priority = options->priority; int min = sched_get_priority_min( SCHED_RR ); int max = sched_get_priority_max( SCHED_RR ); if ( priority < min ) priority = min; else if ( priority > max ) priority = max; stream_.callbackInfo.priority = priority; } #endif stream_.callbackInfo.isRunning = true; result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); pthread_attr_destroy( &attr ); if ( result ) { stream_.callbackInfo.isRunning = false; errorText_ = "RtApiAlsa::error creating callback thread!"; goto error; } } return SUCCESS; error: if ( apiInfo ) { pthread_cond_destroy( &apiInfo->runnable_cv ); if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); delete apiInfo; stream_.apiHandle = 0; } if ( phandle) snd_pcm_close( phandle ); for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } stream_.state = STREAM_CLOSED; return FAILURE; } void RtApiAlsa :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; error( RtAudioError::WARNING ); return; } AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; stream_.callbackInfo.isRunning = false; MUTEX_LOCK( &stream_.mutex ); if ( stream_.state == STREAM_STOPPED ) { apiInfo->runnable = true; pthread_cond_signal( &apiInfo->runnable_cv ); } MUTEX_UNLOCK( &stream_.mutex ); pthread_join( stream_.callbackInfo.thread, NULL ); if ( stream_.state == STREAM_RUNNING ) { stream_.state = STREAM_STOPPED; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) snd_pcm_drop( apiInfo->handles[0] ); if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) snd_pcm_drop( apiInfo->handles[1] ); } if ( apiInfo ) { pthread_cond_destroy( &apiInfo->runnable_cv ); if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); delete apiInfo; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } stream_.mode = UNINITIALIZED; stream_.state = STREAM_CLOSED; } void RtApiAlsa :: startStream() { // This method calls snd_pcm_prepare if the device isn't already in that state. verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } MUTEX_LOCK( &stream_.mutex ); int result = 0; snd_pcm_state_t state; AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { state = snd_pcm_state( handle[0] ); if ( state != SND_PCM_STATE_PREPARED ) { result = snd_pcm_prepare( handle[0] ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } } if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open state = snd_pcm_state( handle[1] ); if ( state != SND_PCM_STATE_PREPARED ) { result = snd_pcm_prepare( handle[1] ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } } stream_.state = STREAM_RUNNING; unlock: apiInfo->runnable = true; pthread_cond_signal( &apiInfo->runnable_cv ); MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiAlsa :: stopStream() { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); int result = 0; AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( apiInfo->synchronized ) result = snd_pcm_drop( handle[0] ); else result = snd_pcm_drain( handle[0] ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { result = snd_pcm_drop( handle[1] ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } unlock: apiInfo->runnable = false; // fixes high CPU usage when stopped MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiAlsa :: abortStream() { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); int result = 0; AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { result = snd_pcm_drop( handle[0] ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { result = snd_pcm_drop( handle[1] ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } unlock: apiInfo->runnable = false; // fixes high CPU usage when stopped MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiAlsa :: callbackEvent() { AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; if ( stream_.state == STREAM_STOPPED ) { MUTEX_LOCK( &stream_.mutex ); while ( !apiInfo->runnable ) pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); if ( stream_.state != STREAM_RUNNING ) { MUTEX_UNLOCK( &stream_.mutex ); return; } MUTEX_UNLOCK( &stream_.mutex ); } if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; error( RtAudioError::WARNING ); return; } int doStopStream = 0; RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { status |= RTAUDIO_OUTPUT_UNDERFLOW; apiInfo->xrun[0] = false; } if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { status |= RTAUDIO_INPUT_OVERFLOW; apiInfo->xrun[1] = false; } doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); if ( doStopStream == 2 ) { abortStream(); return; } MUTEX_LOCK( &stream_.mutex ); // The state might change while waiting on a mutex. if ( stream_.state == STREAM_STOPPED ) goto unlock; int result; char *buffer; int channels; snd_pcm_t **handle; snd_pcm_sframes_t frames; RtAudioFormat format; handle = (snd_pcm_t **) apiInfo->handles; if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { // Setup parameters. if ( stream_.doConvertBuffer[1] ) { buffer = stream_.deviceBuffer; channels = stream_.nDeviceChannels[1]; format = stream_.deviceFormat[1]; } else { buffer = stream_.userBuffer[1]; channels = stream_.nUserChannels[1]; format = stream_.userFormat; } // Read samples from device in interleaved/non-interleaved format. if ( stream_.deviceInterleaved[1] ) result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); else { void *bufs[channels]; size_t offset = stream_.bufferSize * formatBytes( format ); for ( int i=0; ixrun[1] = true; result = snd_pcm_prepare( handle[1] ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } } else { errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } } else { errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } error( RtAudioError::WARNING ); goto tryOutput; } // Do byte swapping if necessary. if ( stream_.doByteSwap[1] ) byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); // Do buffer conversion if necessary. if ( stream_.doConvertBuffer[1] ) convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); // Check stream latency result = snd_pcm_delay( handle[1], &frames ); if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; } tryOutput: if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { // Setup parameters and do buffer conversion if necessary. if ( stream_.doConvertBuffer[0] ) { buffer = stream_.deviceBuffer; convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); channels = stream_.nDeviceChannels[0]; format = stream_.deviceFormat[0]; } else { buffer = stream_.userBuffer[0]; channels = stream_.nUserChannels[0]; format = stream_.userFormat; } // Do byte swapping if necessary. if ( stream_.doByteSwap[0] ) byteSwapBuffer(buffer, stream_.bufferSize * channels, format); // Write samples to device in interleaved/non-interleaved format. if ( stream_.deviceInterleaved[0] ) result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); else { void *bufs[channels]; size_t offset = stream_.bufferSize * formatBytes( format ); for ( int i=0; ixrun[0] = true; result = snd_pcm_prepare( handle[0] ); if ( result < 0 ) { errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } else errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; } else { errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } } else { errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } error( RtAudioError::WARNING ); goto unlock; } // Check stream latency result = snd_pcm_delay( handle[0], &frames ); if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; } unlock: MUTEX_UNLOCK( &stream_.mutex ); RtApi::tickStreamTime(); if ( doStopStream == 1 ) this->stopStream(); } static void *alsaCallbackHandler( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiAlsa *object = (RtApiAlsa *) info->object; bool *isRunning = &info->isRunning; #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) if ( info->doRealtime ) { pthread_t tID = pthread_self(); // ID of this thread sched_param prio = { info->priority }; // scheduling priority of thread pthread_setschedparam( tID, SCHED_RR, &prio ); } #endif while ( *isRunning == true ) { pthread_testcancel(); object->callbackEvent(); } pthread_exit( NULL ); } //******************** End of __LINUX_ALSA__ *********************// #endif #if defined(__LINUX_PULSE__) // Code written by Peter Meerwald, pmeerw@pmeerw.net // and Tristan Matthews. #include #include #include static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, 44100, 48000, 96000, 0}; struct rtaudio_pa_format_mapping_t { RtAudioFormat rtaudio_format; pa_sample_format_t pa_format; }; static const rtaudio_pa_format_mapping_t supported_sampleformats[] = { {RTAUDIO_SINT16, PA_SAMPLE_S16LE}, {RTAUDIO_SINT32, PA_SAMPLE_S32LE}, {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE}, {0, PA_SAMPLE_INVALID}}; struct PulseAudioHandle { pa_simple *s_play; pa_simple *s_rec; pthread_t thread; pthread_cond_t runnable_cv; bool runnable; PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { } }; RtApiPulse::~RtApiPulse() { if ( stream_.state != STREAM_CLOSED ) closeStream(); } unsigned int RtApiPulse::getDeviceCount( void ) { return 1; } RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; info.probed = true; info.name = "PulseAudio"; info.outputChannels = 2; info.inputChannels = 2; info.duplexChannels = 2; info.isDefaultOutput = true; info.isDefaultInput = true; for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) info.sampleRates.push_back( *sr ); info.preferredSampleRate = 48000; info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; return info; } static void *pulseaudio_callback( void * user ) { CallbackInfo *cbi = static_cast( user ); RtApiPulse *context = static_cast( cbi->object ); volatile bool *isRunning = &cbi->isRunning; while ( *isRunning ) { pthread_testcancel(); context->callbackEvent(); } pthread_exit( NULL ); } void RtApiPulse::closeStream( void ) { PulseAudioHandle *pah = static_cast( stream_.apiHandle ); stream_.callbackInfo.isRunning = false; if ( pah ) { MUTEX_LOCK( &stream_.mutex ); if ( stream_.state == STREAM_STOPPED ) { pah->runnable = true; pthread_cond_signal( &pah->runnable_cv ); } MUTEX_UNLOCK( &stream_.mutex ); pthread_join( pah->thread, 0 ); if ( pah->s_play ) { pa_simple_flush( pah->s_play, NULL ); pa_simple_free( pah->s_play ); } if ( pah->s_rec ) pa_simple_free( pah->s_rec ); pthread_cond_destroy( &pah->runnable_cv ); delete pah; stream_.apiHandle = 0; } if ( stream_.userBuffer[0] ) { free( stream_.userBuffer[0] ); stream_.userBuffer[0] = 0; } if ( stream_.userBuffer[1] ) { free( stream_.userBuffer[1] ); stream_.userBuffer[1] = 0; } stream_.state = STREAM_CLOSED; stream_.mode = UNINITIALIZED; } void RtApiPulse::callbackEvent( void ) { PulseAudioHandle *pah = static_cast( stream_.apiHandle ); if ( stream_.state == STREAM_STOPPED ) { MUTEX_LOCK( &stream_.mutex ); while ( !pah->runnable ) pthread_cond_wait( &pah->runnable_cv, &stream_.mutex ); if ( stream_.state != STREAM_RUNNING ) { MUTEX_UNLOCK( &stream_.mutex ); return; } MUTEX_UNLOCK( &stream_.mutex ); } if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " "this shouldn't happen!"; error( RtAudioError::WARNING ); return; } RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT], stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); if ( doStopStream == 2 ) { abortStream(); return; } MUTEX_LOCK( &stream_.mutex ); void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT]; void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT]; if ( stream_.state != STREAM_RUNNING ) goto unlock; int pa_error; size_t bytes; if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( stream_.doConvertBuffer[OUTPUT] ) { convertBuffer( stream_.deviceBuffer, stream_.userBuffer[OUTPUT], stream_.convertInfo[OUTPUT] ); bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize * formatBytes( stream_.deviceFormat[OUTPUT] ); } else bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize * formatBytes( stream_.userFormat ); if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) { errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); } } if ( stream_.mode == INPUT || stream_.mode == DUPLEX) { if ( stream_.doConvertBuffer[INPUT] ) bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize * formatBytes( stream_.deviceFormat[INPUT] ); else bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize * formatBytes( stream_.userFormat ); if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) { errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); } if ( stream_.doConvertBuffer[INPUT] ) { convertBuffer( stream_.userBuffer[INPUT], stream_.deviceBuffer, stream_.convertInfo[INPUT] ); } } unlock: MUTEX_UNLOCK( &stream_.mutex ); RtApi::tickStreamTime(); if ( doStopStream == 1 ) stopStream(); } void RtApiPulse::startStream( void ) { PulseAudioHandle *pah = static_cast( stream_.apiHandle ); if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiPulse::startStream(): the stream is not open!"; error( RtAudioError::INVALID_USE ); return; } if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiPulse::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } MUTEX_LOCK( &stream_.mutex ); stream_.state = STREAM_RUNNING; pah->runnable = true; pthread_cond_signal( &pah->runnable_cv ); MUTEX_UNLOCK( &stream_.mutex ); } void RtApiPulse::stopStream( void ) { PulseAudioHandle *pah = static_cast( stream_.apiHandle ); if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; error( RtAudioError::INVALID_USE ); return; } if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); if ( pah && pah->s_play ) { int pa_error; if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) { errorStream_ << "RtApiPulse::stopStream: error draining output device, " << pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } } stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); } void RtApiPulse::abortStream( void ) { PulseAudioHandle *pah = static_cast( stream_.apiHandle ); if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; error( RtAudioError::INVALID_USE ); return; } if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); if ( pah && pah->s_play ) { int pa_error; if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) { errorStream_ << "RtApiPulse::abortStream: error flushing output device, " << pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); error( RtAudioError::SYSTEM_ERROR ); return; } } stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); } bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ) { PulseAudioHandle *pah = 0; unsigned long bufferBytes = 0; pa_sample_spec ss; if ( device != 0 ) return false; if ( mode != INPUT && mode != OUTPUT ) return false; if ( channels != 1 && channels != 2 ) { errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels."; return false; } ss.channels = channels; if ( firstChannel != 0 ) return false; bool sr_found = false; for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) { if ( sampleRate == *sr ) { sr_found = true; stream_.sampleRate = sampleRate; ss.rate = sampleRate; break; } } if ( !sr_found ) { errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate."; return false; } bool sf_found = 0; for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats; sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) { if ( format == sf->rtaudio_format ) { sf_found = true; stream_.userFormat = sf->rtaudio_format; stream_.deviceFormat[mode] = stream_.userFormat; ss.format = sf->pa_format; break; } } if ( !sf_found ) { // Use internal data format conversion. stream_.userFormat = format; stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; ss.format = PA_SAMPLE_FLOAT32LE; } // Set other stream parameters. if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; else stream_.userInterleaved = true; stream_.deviceInterleaved[mode] = true; stream_.nBuffers = 1; stream_.doByteSwap[mode] = false; stream_.nUserChannels[mode] = channels; stream_.nDeviceChannels[mode] = channels + firstChannel; stream_.channelOffset[mode] = 0; std::string streamName = "RtAudio"; // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] ) stream_.doConvertBuffer[mode] = true; if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) stream_.doConvertBuffer[mode] = true; // Allocate necessary internal buffers. bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory."; goto error; } stream_.bufferSize = *bufferSize; if ( stream_.doConvertBuffer[mode] ) { bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); if ( mode == INPUT ) { if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); if ( bufferBytes <= bytesOut ) makeBuffer = false; } } if ( makeBuffer ) { bufferBytes *= *bufferSize; if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); if ( stream_.deviceBuffer == NULL ) { errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory."; goto error; } } } stream_.device[mode] = device; // Setup the buffer conversion information structure. if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); if ( !stream_.apiHandle ) { PulseAudioHandle *pah = new PulseAudioHandle; if ( !pah ) { errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle."; goto error; } stream_.apiHandle = pah; if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) { errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable."; goto error; } } pah = static_cast( stream_.apiHandle ); int error; if ( options && !options->streamName.empty() ) streamName = options->streamName; switch ( mode ) { case INPUT: pa_buffer_attr buffer_attr; buffer_attr.fragsize = bufferBytes; buffer_attr.maxlength = -1; pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error ); if ( !pah->s_rec ) { errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; goto error; } break; case OUTPUT: pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); if ( !pah->s_play ) { errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; goto error; } break; default: goto error; } if ( stream_.mode == UNINITIALIZED ) stream_.mode = mode; else if ( stream_.mode == mode ) goto error; else stream_.mode = DUPLEX; if ( !stream_.callbackInfo.isRunning ) { stream_.callbackInfo.object = this; stream_.callbackInfo.isRunning = true; if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) { errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; goto error; } } stream_.state = STREAM_STOPPED; return true; error: if ( pah && stream_.callbackInfo.isRunning ) { pthread_cond_destroy( &pah->runnable_cv ); delete pah; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } return FAILURE; } //******************** End of __LINUX_PULSE__ *********************// #endif #if defined(__LINUX_OSS__) #include #include #include #include #include #include #include static void *ossCallbackHandler(void * ptr); // A structure to hold various information related to the OSS API // implementation. struct OssHandle { int id[2]; // device ids bool xrun[2]; bool triggered; pthread_cond_t runnable; OssHandle() :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } }; RtApiOss :: RtApiOss() { // Nothing to do here. } RtApiOss :: ~RtApiOss() { if ( stream_.state != STREAM_CLOSED ) closeStream(); } unsigned int RtApiOss :: getDeviceCount( void ) { int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); if ( mixerfd == -1 ) { errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; error( RtAudioError::WARNING ); return 0; } oss_sysinfo sysinfo; if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { close( mixerfd ); errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; error( RtAudioError::WARNING ); return 0; } close( mixerfd ); return sysinfo.numaudios; } RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; info.probed = false; int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); if ( mixerfd == -1 ) { errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; error( RtAudioError::WARNING ); return info; } oss_sysinfo sysinfo; int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); if ( result == -1 ) { close( mixerfd ); errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; error( RtAudioError::WARNING ); return info; } unsigned nDevices = sysinfo.numaudios; if ( nDevices == 0 ) { close( mixerfd ); errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; error( RtAudioError::INVALID_USE ); return info; } if ( device >= nDevices ) { close( mixerfd ); errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; error( RtAudioError::INVALID_USE ); return info; } oss_audioinfo ainfo; ainfo.dev = device; result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); close( mixerfd ); if ( result == -1 ) { errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Probe channels if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; if ( ainfo.caps & PCM_CAP_DUPLEX ) { if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; } // Probe data formats ... do for input unsigned long mask = ainfo.iformats; if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) info.nativeFormats |= RTAUDIO_SINT16; if ( mask & AFMT_S8 ) info.nativeFormats |= RTAUDIO_SINT8; if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) info.nativeFormats |= RTAUDIO_SINT32; #ifdef AFMT_FLOAT if ( mask & AFMT_FLOAT ) info.nativeFormats |= RTAUDIO_FLOAT32; #endif if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) info.nativeFormats |= RTAUDIO_SINT24; // Check that we have at least one supported format if ( info.nativeFormats == 0 ) { errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); return info; } // Probe the supported sample rates. info.sampleRates.clear(); if ( ainfo.nrates ) { for ( unsigned int i=0; i info.preferredSampleRate ) ) info.preferredSampleRate = SAMPLE_RATES[k]; break; } } } } else { // Check min and max rate values; for ( unsigned int k=0; k= (int) SAMPLE_RATES[k] ) { info.sampleRates.push_back( SAMPLE_RATES[k] ); if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) info.preferredSampleRate = SAMPLE_RATES[k]; } } } if ( info.sampleRates.size() == 0 ) { errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); } else { info.probed = true; info.name = ainfo.name; } return info; } bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ) { int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); if ( mixerfd == -1 ) { errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; return FAILURE; } oss_sysinfo sysinfo; int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); if ( result == -1 ) { close( mixerfd ); errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; return FAILURE; } unsigned nDevices = sysinfo.numaudios; if ( nDevices == 0 ) { // This should not happen because a check is made before this function is called. close( mixerfd ); errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; return FAILURE; } if ( device >= nDevices ) { // This should not happen because a check is made before this function is called. close( mixerfd ); errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; return FAILURE; } oss_audioinfo ainfo; ainfo.dev = device; result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); close( mixerfd ); if ( result == -1 ) { errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; errorText_ = errorStream_.str(); return FAILURE; } // Check if device supports input or output if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { if ( mode == OUTPUT ) errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; else errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; errorText_ = errorStream_.str(); return FAILURE; } int flags = 0; OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( mode == OUTPUT ) flags |= O_WRONLY; else { // mode == INPUT if (stream_.mode == OUTPUT && stream_.device[0] == device) { // We just set the same device for playback ... close and reopen for duplex (OSS only). close( handle->id[0] ); handle->id[0] = 0; if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; errorText_ = errorStream_.str(); return FAILURE; } // Check that the number previously set channels is the same. if ( stream_.nUserChannels[0] != channels ) { errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); return FAILURE; } flags |= O_RDWR; } else flags |= O_RDONLY; } // Set exclusive access if specified. if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; // Try to open the device. int fd; fd = open( ainfo.devnode, flags, 0 ); if ( fd == -1 ) { if ( errno == EBUSY ) errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; else errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); return FAILURE; } // For duplex operation, specifically set this mode (this doesn't seem to work). /* if ( flags | O_RDWR ) { result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); if ( result == -1) { errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); return FAILURE; } } */ // Check the device channel support. stream_.nUserChannels[mode] = channels; if ( ainfo.max_channels < (int)(channels + firstChannel) ) { close( fd ); errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; errorText_ = errorStream_.str(); return FAILURE; } // Set the number of channels. int deviceChannels = channels + firstChannel; result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { close( fd ); errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); return FAILURE; } stream_.nDeviceChannels[mode] = deviceChannels; // Get the data format mask int mask; result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); if ( result == -1 ) { close( fd ); errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; errorText_ = errorStream_.str(); return FAILURE; } // Determine how to set the device format. stream_.userFormat = format; int deviceFormat = -1; stream_.doByteSwap[mode] = false; if ( format == RTAUDIO_SINT8 ) { if ( mask & AFMT_S8 ) { deviceFormat = AFMT_S8; stream_.deviceFormat[mode] = RTAUDIO_SINT8; } } else if ( format == RTAUDIO_SINT16 ) { if ( mask & AFMT_S16_NE ) { deviceFormat = AFMT_S16_NE; stream_.deviceFormat[mode] = RTAUDIO_SINT16; } else if ( mask & AFMT_S16_OE ) { deviceFormat = AFMT_S16_OE; stream_.deviceFormat[mode] = RTAUDIO_SINT16; stream_.doByteSwap[mode] = true; } } else if ( format == RTAUDIO_SINT24 ) { if ( mask & AFMT_S24_NE ) { deviceFormat = AFMT_S24_NE; stream_.deviceFormat[mode] = RTAUDIO_SINT24; } else if ( mask & AFMT_S24_OE ) { deviceFormat = AFMT_S24_OE; stream_.deviceFormat[mode] = RTAUDIO_SINT24; stream_.doByteSwap[mode] = true; } } else if ( format == RTAUDIO_SINT32 ) { if ( mask & AFMT_S32_NE ) { deviceFormat = AFMT_S32_NE; stream_.deviceFormat[mode] = RTAUDIO_SINT32; } else if ( mask & AFMT_S32_OE ) { deviceFormat = AFMT_S32_OE; stream_.deviceFormat[mode] = RTAUDIO_SINT32; stream_.doByteSwap[mode] = true; } } if ( deviceFormat == -1 ) { // The user requested format is not natively supported by the device. if ( mask & AFMT_S16_NE ) { deviceFormat = AFMT_S16_NE; stream_.deviceFormat[mode] = RTAUDIO_SINT16; } else if ( mask & AFMT_S32_NE ) { deviceFormat = AFMT_S32_NE; stream_.deviceFormat[mode] = RTAUDIO_SINT32; } else if ( mask & AFMT_S24_NE ) { deviceFormat = AFMT_S24_NE; stream_.deviceFormat[mode] = RTAUDIO_SINT24; } else if ( mask & AFMT_S16_OE ) { deviceFormat = AFMT_S16_OE; stream_.deviceFormat[mode] = RTAUDIO_SINT16; stream_.doByteSwap[mode] = true; } else if ( mask & AFMT_S32_OE ) { deviceFormat = AFMT_S32_OE; stream_.deviceFormat[mode] = RTAUDIO_SINT32; stream_.doByteSwap[mode] = true; } else if ( mask & AFMT_S24_OE ) { deviceFormat = AFMT_S24_OE; stream_.deviceFormat[mode] = RTAUDIO_SINT24; stream_.doByteSwap[mode] = true; } else if ( mask & AFMT_S8) { deviceFormat = AFMT_S8; stream_.deviceFormat[mode] = RTAUDIO_SINT8; } } if ( stream_.deviceFormat[mode] == 0 ) { // This really shouldn't happen ... close( fd ); errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; errorText_ = errorStream_.str(); return FAILURE; } // Set the data format. int temp = deviceFormat; result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); if ( result == -1 || deviceFormat != temp ) { close( fd ); errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); return FAILURE; } // Attempt to set the buffer size. According to OSS, the minimum // number of buffers is two. The supposed minimum buffer size is 16 // bytes, so that will be our lower bound. The argument to this // call is in the form 0xMMMMSSSS (hex), where the buffer size (in // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. // We'll check the actual value used near the end of the setup // procedure. int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; if ( ossBufferBytes < 16 ) ossBufferBytes = 16; int buffers = 0; if ( options ) buffers = options->numberOfBuffers; if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; if ( buffers < 2 ) buffers = 3; temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); if ( result == -1 ) { close( fd ); errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); return FAILURE; } stream_.nBuffers = buffers; // Save buffer size (in sample frames). *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); stream_.bufferSize = *bufferSize; // Set the sample rate. int srate = sampleRate; result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); if ( result == -1 ) { close( fd ); errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); return FAILURE; } // Verify the sample rate setup worked. if ( abs( srate - (int)sampleRate ) > 100 ) { close( fd ); errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; errorText_ = errorStream_.str(); return FAILURE; } stream_.sampleRate = sampleRate; if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { // We're doing duplex setup here. stream_.deviceFormat[0] = stream_.deviceFormat[1]; stream_.nDeviceChannels[0] = deviceChannels; } // Set interleaving parameters. stream_.userInterleaved = true; stream_.deviceInterleaved[mode] = true; if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; // Set flags for buffer conversion stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] ) stream_.doConvertBuffer[mode] = true; if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) stream_.doConvertBuffer[mode] = true; if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && stream_.nUserChannels[mode] > 1 ) stream_.doConvertBuffer[mode] = true; // Allocate the stream handles if necessary and then save. if ( stream_.apiHandle == 0 ) { try { handle = new OssHandle; } catch ( std::bad_alloc& ) { errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; goto error; } if ( pthread_cond_init( &handle->runnable, NULL ) ) { errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; goto error; } stream_.apiHandle = (void *) handle; } else { handle = (OssHandle *) stream_.apiHandle; } handle->id[mode] = fd; // Allocate necessary internal buffers. unsigned long bufferBytes; bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; goto error; } if ( stream_.doConvertBuffer[mode] ) { bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); if ( mode == INPUT ) { if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); if ( bufferBytes <= bytesOut ) makeBuffer = false; } } if ( makeBuffer ) { bufferBytes *= *bufferSize; if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); if ( stream_.deviceBuffer == NULL ) { errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; goto error; } } } stream_.device[mode] = device; stream_.state = STREAM_STOPPED; // Setup the buffer conversion information structure. if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); // Setup thread if necessary. if ( stream_.mode == OUTPUT && mode == INPUT ) { // We had already set up an output stream. stream_.mode = DUPLEX; if ( stream_.device[0] == device ) handle->id[0] = fd; } else { stream_.mode = mode; // Setup callback thread. stream_.callbackInfo.object = (void *) this; // Set the thread attributes for joinable and realtime scheduling // priority. The higher priority will only take affect if the // program is run as root or suid. pthread_attr_t attr; pthread_attr_init( &attr ); pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { struct sched_param param; int priority = options->priority; int min = sched_get_priority_min( SCHED_RR ); int max = sched_get_priority_max( SCHED_RR ); if ( priority < min ) priority = min; else if ( priority > max ) priority = max; param.sched_priority = priority; pthread_attr_setschedparam( &attr, ¶m ); pthread_attr_setschedpolicy( &attr, SCHED_RR ); } else pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #else pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #endif stream_.callbackInfo.isRunning = true; result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); pthread_attr_destroy( &attr ); if ( result ) { stream_.callbackInfo.isRunning = false; errorText_ = "RtApiOss::error creating callback thread!"; goto error; } } return SUCCESS; error: if ( handle ) { pthread_cond_destroy( &handle->runnable ); if ( handle->id[0] ) close( handle->id[0] ); if ( handle->id[1] ) close( handle->id[1] ); delete handle; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } return FAILURE; } void RtApiOss :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiOss::closeStream(): no open stream to close!"; error( RtAudioError::WARNING ); return; } OssHandle *handle = (OssHandle *) stream_.apiHandle; stream_.callbackInfo.isRunning = false; MUTEX_LOCK( &stream_.mutex ); if ( stream_.state == STREAM_STOPPED ) pthread_cond_signal( &handle->runnable ); MUTEX_UNLOCK( &stream_.mutex ); pthread_join( stream_.callbackInfo.thread, NULL ); if ( stream_.state == STREAM_RUNNING ) { if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); else ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); stream_.state = STREAM_STOPPED; } if ( handle ) { pthread_cond_destroy( &handle->runnable ); if ( handle->id[0] ) close( handle->id[0] ); if ( handle->id[1] ) close( handle->id[1] ); delete handle; stream_.apiHandle = 0; } for ( int i=0; i<2; i++ ) { if ( stream_.userBuffer[i] ) { free( stream_.userBuffer[i] ); stream_.userBuffer[i] = 0; } } if ( stream_.deviceBuffer ) { free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } stream_.mode = UNINITIALIZED; stream_.state = STREAM_CLOSED; } void RtApiOss :: startStream() { verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiOss::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } MUTEX_LOCK( &stream_.mutex ); stream_.state = STREAM_RUNNING; // No need to do anything else here ... OSS automatically starts // when fed samples. MUTEX_UNLOCK( &stream_.mutex ); OssHandle *handle = (OssHandle *) stream_.apiHandle; pthread_cond_signal( &handle->runnable ); } void RtApiOss :: stopStream() { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } MUTEX_LOCK( &stream_.mutex ); // The state might change while waiting on a mutex. if ( stream_.state == STREAM_STOPPED ) { MUTEX_UNLOCK( &stream_.mutex ); return; } int result = 0; OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { // Flush the output with zeros a few times. char *buffer; int samples; RtAudioFormat format; if ( stream_.doConvertBuffer[0] ) { buffer = stream_.deviceBuffer; samples = stream_.bufferSize * stream_.nDeviceChannels[0]; format = stream_.deviceFormat[0]; } else { buffer = stream_.userBuffer[0]; samples = stream_.bufferSize * stream_.nUserChannels[0]; format = stream_.userFormat; } memset( buffer, 0, samples * formatBytes(format) ); for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); if ( result == -1 ) { errorText_ = "RtApiOss::stopStream: audio write error."; error( RtAudioError::WARNING ); } } result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); if ( result == -1 ) { errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; errorText_ = errorStream_.str(); goto unlock; } handle->triggered = false; } if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); if ( result == -1 ) { errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; errorText_ = errorStream_.str(); goto unlock; } } unlock: stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); if ( result != -1 ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiOss :: abortStream() { verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); return; } MUTEX_LOCK( &stream_.mutex ); // The state might change while waiting on a mutex. if ( stream_.state == STREAM_STOPPED ) { MUTEX_UNLOCK( &stream_.mutex ); return; } int result = 0; OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); if ( result == -1 ) { errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; errorText_ = errorStream_.str(); goto unlock; } handle->triggered = false; } if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); if ( result == -1 ) { errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; errorText_ = errorStream_.str(); goto unlock; } } unlock: stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); if ( result != -1 ) return; error( RtAudioError::SYSTEM_ERROR ); } void RtApiOss :: callbackEvent() { OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.state == STREAM_STOPPED ) { MUTEX_LOCK( &stream_.mutex ); pthread_cond_wait( &handle->runnable, &stream_.mutex ); if ( stream_.state != STREAM_RUNNING ) { MUTEX_UNLOCK( &stream_.mutex ); return; } MUTEX_UNLOCK( &stream_.mutex ); } if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; error( RtAudioError::WARNING ); return; } // Invoke user callback to get fresh output data. int doStopStream = 0; RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; if ( stream_.mode != INPUT && handle->xrun[0] == true ) { status |= RTAUDIO_OUTPUT_UNDERFLOW; handle->xrun[0] = false; } if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { status |= RTAUDIO_INPUT_OVERFLOW; handle->xrun[1] = false; } doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); if ( doStopStream == 2 ) { this->abortStream(); return; } MUTEX_LOCK( &stream_.mutex ); // The state might change while waiting on a mutex. if ( stream_.state == STREAM_STOPPED ) goto unlock; int result; char *buffer; int samples; RtAudioFormat format; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { // Setup parameters and do buffer conversion if necessary. if ( stream_.doConvertBuffer[0] ) { buffer = stream_.deviceBuffer; convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); samples = stream_.bufferSize * stream_.nDeviceChannels[0]; format = stream_.deviceFormat[0]; } else { buffer = stream_.userBuffer[0]; samples = stream_.bufferSize * stream_.nUserChannels[0]; format = stream_.userFormat; } // Do byte swapping if necessary. if ( stream_.doByteSwap[0] ) byteSwapBuffer( buffer, samples, format ); if ( stream_.mode == DUPLEX && handle->triggered == false ) { int trig = 0; ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); result = write( handle->id[0], buffer, samples * formatBytes(format) ); trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); handle->triggered = true; } else // Write samples to device. result = write( handle->id[0], buffer, samples * formatBytes(format) ); if ( result == -1 ) { // We'll assume this is an underrun, though there isn't a // specific means for determining that. handle->xrun[0] = true; errorText_ = "RtApiOss::callbackEvent: audio write error."; error( RtAudioError::WARNING ); // Continue on to input section. } } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { // Setup parameters. if ( stream_.doConvertBuffer[1] ) { buffer = stream_.deviceBuffer; samples = stream_.bufferSize * stream_.nDeviceChannels[1]; format = stream_.deviceFormat[1]; } else { buffer = stream_.userBuffer[1]; samples = stream_.bufferSize * stream_.nUserChannels[1]; format = stream_.userFormat; } // Read samples from device. result = read( handle->id[1], buffer, samples * formatBytes(format) ); if ( result == -1 ) { // We'll assume this is an overrun, though there isn't a // specific means for determining that. handle->xrun[1] = true; errorText_ = "RtApiOss::callbackEvent: audio read error."; error( RtAudioError::WARNING ); goto unlock; } // Do byte swapping if necessary. if ( stream_.doByteSwap[1] ) byteSwapBuffer( buffer, samples, format ); // Do buffer conversion if necessary. if ( stream_.doConvertBuffer[1] ) convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); } unlock: MUTEX_UNLOCK( &stream_.mutex ); RtApi::tickStreamTime(); if ( doStopStream == 1 ) this->stopStream(); } static void *ossCallbackHandler( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiOss *object = (RtApiOss *) info->object; bool *isRunning = &info->isRunning; while ( *isRunning == true ) { pthread_testcancel(); object->callbackEvent(); } pthread_exit( NULL ); } //******************** End of __LINUX_OSS__ *********************// #endif // *************************************************** // // // Protected common (OS-independent) RtAudio methods. // // *************************************************** // // This method can be modified to control the behavior of error // message printing. void RtApi :: error( RtAudioError::Type type ) { errorStream_.str(""); // clear the ostringstream RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback; if ( errorCallback ) { // abortStream() can generate new error messages. Ignore them. Just keep original one. if ( firstErrorOccurred_ ) return; firstErrorOccurred_ = true; const std::string errorMessage = errorText_; if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) { stream_.callbackInfo.isRunning = false; // exit from the thread abortStream(); } errorCallback( type, errorMessage ); firstErrorOccurred_ = false; return; } if ( type == RtAudioError::WARNING && showWarnings_ == true ) std::cerr << '\n' << errorText_ << "\n\n"; else if ( type != RtAudioError::WARNING ) throw( RtAudioError( errorText_, type ) ); } void RtApi :: verifyStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApi:: a stream is not open!"; error( RtAudioError::INVALID_USE ); } } void RtApi :: clearStreamInfo() { stream_.mode = UNINITIALIZED; stream_.state = STREAM_CLOSED; stream_.sampleRate = 0; stream_.bufferSize = 0; stream_.nBuffers = 0; stream_.userFormat = 0; stream_.userInterleaved = true; stream_.streamTime = 0.0; stream_.apiHandle = 0; stream_.deviceBuffer = 0; stream_.callbackInfo.callback = 0; stream_.callbackInfo.userData = 0; stream_.callbackInfo.isRunning = false; stream_.callbackInfo.errorCallback = 0; for ( int i=0; i<2; i++ ) { stream_.device[i] = 11111; stream_.doConvertBuffer[i] = false; stream_.deviceInterleaved[i] = true; stream_.doByteSwap[i] = false; stream_.nUserChannels[i] = 0; stream_.nDeviceChannels[i] = 0; stream_.channelOffset[i] = 0; stream_.deviceFormat[i] = 0; stream_.latency[i] = 0; stream_.userBuffer[i] = 0; stream_.convertInfo[i].channels = 0; stream_.convertInfo[i].inJump = 0; stream_.convertInfo[i].outJump = 0; stream_.convertInfo[i].inFormat = 0; stream_.convertInfo[i].outFormat = 0; stream_.convertInfo[i].inOffset.clear(); stream_.convertInfo[i].outOffset.clear(); } } unsigned int RtApi :: formatBytes( RtAudioFormat format ) { if ( format == RTAUDIO_SINT16 ) return 2; else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 ) return 4; else if ( format == RTAUDIO_FLOAT64 ) return 8; else if ( format == RTAUDIO_SINT24 ) return 3; else if ( format == RTAUDIO_SINT8 ) return 1; errorText_ = "RtApi::formatBytes: undefined format."; error( RtAudioError::WARNING ); return 0; } void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) { if ( mode == INPUT ) { // convert device to user buffer stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; stream_.convertInfo[mode].outFormat = stream_.userFormat; } else { // convert user to device buffer stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; stream_.convertInfo[mode].inFormat = stream_.userFormat; stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; } if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; else stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; // Set up the interleave/deinterleave offsets. if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || ( mode == INPUT && stream_.userInterleaved ) ) { for ( int k=0; k 0 ) { if ( stream_.deviceInterleaved[mode] ) { if ( mode == OUTPUT ) { for ( int k=0; k> 8); //out[info.outOffset[j]] >>= 8; } in += info.inJump; out += info.outJump; } } else if (info.inFormat == RTAUDIO_FLOAT32) { Float32 *in = (Float32 *)inBuffer; for (unsigned int i=0; i> 8); } in += info.inJump; out += info.outJump; } } else if (info.inFormat == RTAUDIO_SINT32) { Int32 *in = (Int32 *)inBuffer; for (unsigned int i=0; i> 16) & 0x0000ffff); } in += info.inJump; out += info.outJump; } } else if (info.inFormat == RTAUDIO_FLOAT32) { Float32 *in = (Float32 *)inBuffer; for (unsigned int i=0; i> 8) & 0x00ff); } in += info.inJump; out += info.outJump; } } else if (info.inFormat == RTAUDIO_SINT24) { Int24 *in = (Int24 *)inBuffer; for (unsigned int i=0; i> 16); } in += info.inJump; out += info.outJump; } } else if (info.inFormat == RTAUDIO_SINT32) { Int32 *in = (Int32 *)inBuffer; for (unsigned int i=0; i> 24) & 0x000000ff); } in += info.inJump; out += info.outJump; } } else if (info.inFormat == RTAUDIO_FLOAT32) { Float32 *in = (Float32 *)inBuffer; for (unsigned int i=0; i>8) | (x<<8); } //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) { char val; char *ptr; ptr = buffer; if ( format == RTAUDIO_SINT16 ) { for ( unsigned int i=0; i #include #include #include /*! \typedef typedef unsigned long RtAudioFormat; \brief RtAudio data format type. Support for signed integers and floats. Audio data fed to/from an RtAudio stream is assumed to ALWAYS be in host byte order. The internal routines will automatically take care of any necessary byte-swapping between the host format and the soundcard. Thus, endian-ness is not a concern in the following format definitions. - \e RTAUDIO_SINT8: 8-bit signed integer. - \e RTAUDIO_SINT16: 16-bit signed integer. - \e RTAUDIO_SINT24: 24-bit signed integer. - \e RTAUDIO_SINT32: 32-bit signed integer. - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0. - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0. */ typedef unsigned long RtAudioFormat; static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer. static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer. static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer. static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer. static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0. static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0. /*! \typedef typedef unsigned long RtAudioStreamFlags; \brief RtAudio stream option flags. The following flags can be OR'ed together to allow a client to make changes to the default stream behavior: - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). - \e RTAUDIO_JACK_DONT_CONNECT: Do not automatically connect ports (JACK only). By default, RtAudio streams pass and receive audio data from the client in an interleaved format. By passing the RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio data will instead be presented in non-interleaved buffers. In this case, each buffer argument in the RtAudioCallback function will point to a single array of data, with \c nFrames samples for each channel concatenated back-to-back. For example, the first sample of data for the second channel would be located at index \c nFrames (assuming the \c buffer pointer was recast to the correct data type for the stream). Certain audio APIs offer a number of parameters that influence the I/O latency of a stream. By default, RtAudio will attempt to set these parameters internally for robust (glitch-free) performance (though some APIs, like Windows Direct Sound, make this difficult). By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() function, internal stream settings will be influenced in an attempt to minimize stream latency, though possibly at the expense of stream performance. If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to open the input and/or output stream device(s) for exclusive use. Note that this is not possible with all supported audio APIs. If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt to select realtime scheduling (round-robin) for the callback thread. If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to open the "default" PCM device when using the ALSA API. Note that this will override any specified input or output device id. If the RTAUDIO_JACK_DONT_CONNECT flag is set, RtAudio will not attempt to automatically connect the ports of the client to the audio device. */ typedef unsigned int RtAudioStreamFlags; static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved). static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency. static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread. static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only). static const RtAudioStreamFlags RTAUDIO_JACK_DONT_CONNECT = 0x20; // Do not automatically connect ports (JACK only). /*! \typedef typedef unsigned long RtAudioStreamStatus; \brief RtAudio stream status (over- or underflow) flags. Notification of a stream over- or underflow is indicated by a non-zero stream \c status argument in the RtAudioCallback function. The stream status can be one of the following two options, depending on whether the stream is open for output and/or input: - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver. - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound. */ typedef unsigned int RtAudioStreamStatus; static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver. static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound. //! RtAudio callback function prototype. /*! All RtAudio clients must create a function of type RtAudioCallback to read and/or write data from/to the audio stream. When the underlying audio system is ready for new input or output data, this function will be invoked. \param outputBuffer For output (or duplex) streams, the client should write \c nFrames of audio sample frames into this buffer. This argument should be recast to the datatype specified when the stream was opened. For input-only streams, this argument will be NULL. \param inputBuffer For input (or duplex) streams, this buffer will hold \c nFrames of input audio sample frames. This argument should be recast to the datatype specified when the stream was opened. For output-only streams, this argument will be NULL. \param nFrames The number of sample frames of input or output data in the buffers. The actual buffer size in bytes is dependent on the data type and number of channels in use. \param streamTime The number of seconds that have elapsed since the stream was started. \param status If non-zero, this argument indicates a data overflow or underflow condition for the stream. The particular condition can be determined by comparison with the RtAudioStreamStatus flags. \param userData A pointer to optional data provided by the client when opening the stream (default = NULL). To continue normal stream operation, the RtAudioCallback function should return a value of zero. To stop the stream and drain the output buffer, the function should return a value of one. To abort the stream immediately, the client should return a value of two. */ typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status, void *userData ); /************************************************************************/ /*! \class RtAudioError \brief Exception handling class for RtAudio. The RtAudioError class is quite simple but it does allow errors to be "caught" by RtAudioError::Type. See the RtAudio documentation to know which methods can throw an RtAudioError. */ /************************************************************************/ class RtAudioError : public std::runtime_error { public: //! Defined RtAudioError types. enum Type { WARNING, /*!< A non-critical error. */ DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ UNSPECIFIED, /*!< The default, unspecified error type. */ NO_DEVICES_FOUND, /*!< No devices found on system. */ INVALID_DEVICE, /*!< An invalid device ID was specified. */ MEMORY_ERROR, /*!< An error occured during memory allocation. */ INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ INVALID_USE, /*!< The function was called incorrectly. */ DRIVER_ERROR, /*!< A system driver error occured. */ SYSTEM_ERROR, /*!< A system error occured. */ THREAD_ERROR /*!< A thread error occured. */ }; //! The constructor. RtAudioError( const std::string& message, Type type = RtAudioError::UNSPECIFIED ) : std::runtime_error(message), type_(type) {} //! Prints thrown error message to stderr. virtual void printMessage( void ) const { std::cerr << '\n' << what() << "\n\n"; } //! Returns the thrown error message type. virtual const Type& getType(void) const { return type_; } //! Returns the thrown error message string. virtual const std::string getMessage(void) const { return std::string(what()); } protected: Type type_; }; //! RtAudio error callback function prototype. /*! \param type Type of error. \param errorText Error description. */ typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText ); // **************************************************************** // // // RtAudio class declaration. // // RtAudio is a "controller" used to select an available audio i/o // interface. It presents a common API for the user to call but all // functionality is implemented by the class RtApi and its // subclasses. RtAudio creates an instance of an RtApi subclass // based on the user's API choice. If no choice is made, RtAudio // attempts to make a "logical" API selection. // // **************************************************************** // class RtApi; class RtAudio { public: //! Audio API specifier arguments. enum Api { UNSPECIFIED, /*!< Search for a working compiled API. */ LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */ LINUX_PULSE, /*!< The Linux PulseAudio API. */ LINUX_OSS, /*!< The Linux Open Sound System API. */ UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */ WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ WINDOWS_DS, /*!< The Microsoft Direct Sound API. */ RTAUDIO_DUMMY /*!< A compilable but non-functional API. */ }; //! The public device information structure for returning queried values. struct DeviceInfo { bool probed; /*!< true if the device capabilities were successfully probed. */ std::string name; /*!< Character string device identifier. */ unsigned int outputChannels; /*!< Maximum output channels supported by device. */ unsigned int inputChannels; /*!< Maximum input channels supported by device. */ unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */ bool isDefaultOutput; /*!< true if this is the default output device. */ bool isDefaultInput; /*!< true if this is the default input device. */ std::vector sampleRates; /*!< Supported sample rates (queried from list of standard rates). */ unsigned int preferredSampleRate; /*!< Preferred sample rate, eg. for WASAPI the system sample rate. */ RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */ // Default constructor. DeviceInfo() :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0), isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), nativeFormats(0) {} }; //! The structure for specifying input or ouput stream parameters. struct StreamParameters { unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */ unsigned int nChannels; /*!< Number of channels. */ unsigned int firstChannel; /*!< First channel index on device (default = 0). */ // Default constructor. StreamParameters() : deviceId(0), nChannels(0), firstChannel(0) {} }; //! The structure for specifying stream options. /*! The following flags can be OR'ed together to allow a client to make changes to the default stream behavior: - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread. - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). By default, RtAudio streams pass and receive audio data from the client in an interleaved format. By passing the RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio data will instead be presented in non-interleaved buffers. In this case, each buffer argument in the RtAudioCallback function will point to a single array of data, with \c nFrames samples for each channel concatenated back-to-back. For example, the first sample of data for the second channel would be located at index \c nFrames (assuming the \c buffer pointer was recast to the correct data type for the stream). Certain audio APIs offer a number of parameters that influence the I/O latency of a stream. By default, RtAudio will attempt to set these parameters internally for robust (glitch-free) performance (though some APIs, like Windows Direct Sound, make this difficult). By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() function, internal stream settings will be influenced in an attempt to minimize stream latency, though possibly at the expense of stream performance. If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to open the input and/or output stream device(s) for exclusive use. Note that this is not possible with all supported audio APIs. If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt to select realtime scheduling (round-robin) for the callback thread. The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME flag is set. It defines the thread's realtime priority. If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to open the "default" PCM device when using the ALSA API. Note that this will override any specified input or output device id. The \c numberOfBuffers parameter can be used to control stream latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs only. A value of two is usually the smallest allowed. Larger numbers can potentially result in more robust stream performance, though likely at the cost of stream latency. The value set by the user is replaced during execution of the RtAudio::openStream() function by the value actually used by the system. The \c streamName parameter can be used to set the client name when using the Jack API. By default, the client name is set to RtApiJack. However, if you wish to create multiple instances of RtAudio with Jack, each instance must have a unique client name. */ struct StreamOptions { RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */ unsigned int numberOfBuffers; /*!< Number of stream buffers. */ std::string streamName; /*!< A stream name (currently used only in Jack). */ int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */ // Default constructor. StreamOptions() : flags(0), numberOfBuffers(0), priority(0) {} }; //! A static function to determine the current RtAudio version. static std::string getVersion( void ); //! A static function to determine the available compiled audio APIs. /*! The values returned in the std::vector can be compared against the enumerated list values. Note that there can be more than one API compiled for certain operating systems. */ static void getCompiledApi( std::vector &apis ); //! The class constructor. /*! The constructor performs minor initialization tasks. An exception can be thrown if no API support is compiled. If no API argument is specified and multiple API support has been compiled, the default order of use is JACK, ALSA, OSS (Linux systems) and ASIO, DS (Windows systems). */ RtAudio( RtAudio::Api api=UNSPECIFIED ); //! The destructor. /*! If a stream is running or open, it will be stopped and closed automatically. */ ~RtAudio(); //! Returns the audio API specifier for the current instance of RtAudio. RtAudio::Api getCurrentApi( void ); //! A public function that queries for the number of audio devices available. /*! This function performs a system query of available devices each time it is called, thus supporting devices connected \e after instantiation. If a system error occurs during processing, a warning will be issued. */ unsigned int getDeviceCount( void ); //! Return an RtAudio::DeviceInfo structure for a specified device number. /*! Any device integer between 0 and getDeviceCount() - 1 is valid. If an invalid argument is provided, an RtAudioError (type = INVALID_USE) will be thrown. If a device is busy or otherwise unavailable, the structure member "probed" will have a value of "false" and all other members are undefined. If the specified device is the current default input or output device, the corresponding "isDefault" member will have a value of "true". */ RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); //! A function that returns the index of the default output device. /*! If the underlying audio API does not provide a "default device", or if no devices are available, the return value will be 0. Note that this is a valid device identifier and it is the client's responsibility to verify that a device is available before attempting to open a stream. */ unsigned int getDefaultOutputDevice( void ); //! A function that returns the index of the default input device. /*! If the underlying audio API does not provide a "default device", or if no devices are available, the return value will be 0. Note that this is a valid device identifier and it is the client's responsibility to verify that a device is available before attempting to open a stream. */ unsigned int getDefaultInputDevice( void ); //! A public function for opening a stream with the specified parameters. /*! An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be opened with the specified parameters or an error occurs during processing. An RtAudioError (type = INVALID_USE) is thrown if any invalid device ID or channel number parameters are specified. \param outputParameters Specifies output stream parameters to use when opening a stream, including a device ID, number of channels, and starting channel number. For input-only streams, this argument should be NULL. The device ID is an index value between 0 and getDeviceCount() - 1. \param inputParameters Specifies input stream parameters to use when opening a stream, including a device ID, number of channels, and starting channel number. For output-only streams, this argument should be NULL. The device ID is an index value between 0 and getDeviceCount() - 1. \param format An RtAudioFormat specifying the desired sample data format. \param sampleRate The desired sample rate (sample frames per second). \param *bufferFrames A pointer to a value indicating the desired internal buffer size in sample frames. The actual value used by the device is returned via the same pointer. A value of zero can be specified, in which case the lowest allowable value is determined. \param callback A client-defined function that will be invoked when input data is available and/or output data is needed. \param userData An optional pointer to data that can be accessed from within the callback function. \param options An optional pointer to a structure containing various global stream options, including a list of OR'ed RtAudioStreamFlags and a suggested number of stream buffers that can be used to control stream latency. More buffers typically result in more robust performance, though at a cost of greater latency. If a value of zero is specified, a system-specific median value is chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the lowest allowable value is used. The actual value used is returned via the structure argument. The parameter is API dependent. \param errorCallback A client-defined function that will be invoked when an error has occured. */ void openStream( RtAudio::StreamParameters *outputParameters, RtAudio::StreamParameters *inputParameters, RtAudioFormat format, unsigned int sampleRate, unsigned int *bufferFrames, RtAudioCallback callback, void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL ); //! A function that closes a stream and frees any associated stream memory. /*! If a stream is not open, this function issues a warning and returns (no exception is thrown). */ void closeStream( void ); //! A function that starts a stream. /*! An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs during processing. An RtAudioError (type = INVALID_USE) is thrown if a stream is not open. A warning is issued if the stream is already running. */ void startStream( void ); //! Stop a stream, allowing any samples remaining in the output queue to be played. /*! An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs during processing. An RtAudioError (type = INVALID_USE) is thrown if a stream is not open. A warning is issued if the stream is already stopped. */ void stopStream( void ); //! Stop a stream, discarding any samples remaining in the input/output queue. /*! An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs during processing. An RtAudioError (type = INVALID_USE) is thrown if a stream is not open. A warning is issued if the stream is already stopped. */ void abortStream( void ); //! Returns true if a stream is open and false if not. bool isStreamOpen( void ) const; //! Returns true if the stream is running and false if it is stopped or not open. bool isStreamRunning( void ) const; //! Returns the number of elapsed seconds since the stream was started. /*! If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. */ double getStreamTime( void ); //! Set the stream time to a time in seconds greater than or equal to 0.0. /*! If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. */ void setStreamTime( double time ); //! Returns the internal stream latency in sample frames. /*! The stream latency refers to delay in audio input and/or output caused by internal buffering by the audio system and/or hardware. For duplex streams, the returned value will represent the sum of the input and output latencies. If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. If the API does not report latency, the return value will be zero. */ long getStreamLatency( void ); //! Returns actual sample rate in use by the stream. /*! On some systems, the sample rate used may be slightly different than that specified in the stream parameters. If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. */ unsigned int getStreamSampleRate( void ); //! Specify whether warning messages should be printed to stderr. void showWarnings( bool value = true ); protected: void openRtApi( RtAudio::Api api ); RtApi *rtapi_; }; // Operating system dependent thread functionality. #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) #ifndef NOMINMAX #define NOMINMAX #endif #include #include typedef uintptr_t ThreadHandle; typedef CRITICAL_SECTION StreamMutex; #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) // Using pthread library for various flavors of unix. #include typedef pthread_t ThreadHandle; typedef pthread_mutex_t StreamMutex; #else // Setup for "dummy" behavior #define __RTAUDIO_DUMMY__ typedef int ThreadHandle; typedef int StreamMutex; #endif // This global structure type is used to pass callback information // between the private RtAudio stream structure and global callback // handling functions. struct CallbackInfo { void *object; // Used as a "this" pointer. ThreadHandle thread; void *callback; void *userData; void *errorCallback; void *apiInfo; // void pointer for API specific callback information bool isRunning; bool doRealtime; int priority; // Default constructor. CallbackInfo() :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false), priority(0) {} }; // **************************************************************** // // // RtApi class declaration. // // Subclasses of RtApi contain all API- and OS-specific code necessary // to fully implement the RtAudio API. // // Note that RtApi is an abstract base class and cannot be // explicitly instantiated. The class RtAudio will create an // instance of an RtApi subclass (RtApiOss, RtApiAlsa, // RtApiJack, RtApiCore, RtApiDs, or RtApiAsio). // // **************************************************************** // #pragma pack(push, 1) class S24 { protected: unsigned char c3[3]; public: S24() {} S24& operator = ( const int& i ) { c3[0] = (i & 0x000000ff); c3[1] = (i & 0x0000ff00) >> 8; c3[2] = (i & 0x00ff0000) >> 16; return *this; } S24( const S24& v ) { *this = v; } S24( const double& d ) { *this = (int) d; } S24( const float& f ) { *this = (int) f; } S24( const signed short& s ) { *this = (int) s; } S24( const char& c ) { *this = (int) c; } int asInt() { int i = c3[0] | (c3[1] << 8) | (c3[2] << 16); if (i & 0x800000) i |= ~0xffffff; return i; } }; #pragma pack(pop) #if defined( HAVE_GETTIMEOFDAY ) #include #endif #include class RtApi { public: RtApi(); virtual ~RtApi(); virtual RtAudio::Api getCurrentApi( void ) = 0; virtual unsigned int getDeviceCount( void ) = 0; virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; virtual unsigned int getDefaultInputDevice( void ); virtual unsigned int getDefaultOutputDevice( void ); void openStream( RtAudio::StreamParameters *outputParameters, RtAudio::StreamParameters *inputParameters, RtAudioFormat format, unsigned int sampleRate, unsigned int *bufferFrames, RtAudioCallback callback, void *userData, RtAudio::StreamOptions *options, RtAudioErrorCallback errorCallback ); virtual void closeStream( void ); virtual void startStream( void ) = 0; virtual void stopStream( void ) = 0; virtual void abortStream( void ) = 0; long getStreamLatency( void ); unsigned int getStreamSampleRate( void ); virtual double getStreamTime( void ); virtual void setStreamTime( double time ); bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; } bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; } void showWarnings( bool value ) { showWarnings_ = value; } protected: static const unsigned int MAX_SAMPLE_RATES; static const unsigned int SAMPLE_RATES[]; enum { FAILURE, SUCCESS }; enum StreamState { STREAM_STOPPED, STREAM_STOPPING, STREAM_RUNNING, STREAM_CLOSED = -50 }; enum StreamMode { OUTPUT, INPUT, DUPLEX, UNINITIALIZED = -75 }; // A protected structure used for buffer conversion. struct ConvertInfo { int channels; int inJump, outJump; RtAudioFormat inFormat, outFormat; std::vector inOffset; std::vector outOffset; }; // A protected structure for audio streams. struct RtApiStream { unsigned int device[2]; // Playback and record, respectively. void *apiHandle; // void pointer for API specific stream handle information StreamMode mode; // OUTPUT, INPUT, or DUPLEX. StreamState state; // STOPPED, RUNNING, or CLOSED char *userBuffer[2]; // Playback and record, respectively. char *deviceBuffer; bool doConvertBuffer[2]; // Playback and record, respectively. bool userInterleaved; bool deviceInterleaved[2]; // Playback and record, respectively. bool doByteSwap[2]; // Playback and record, respectively. unsigned int sampleRate; unsigned int bufferSize; unsigned int nBuffers; unsigned int nUserChannels[2]; // Playback and record, respectively. unsigned int nDeviceChannels[2]; // Playback and record channels, respectively. unsigned int channelOffset[2]; // Playback and record, respectively. unsigned long latency[2]; // Playback and record, respectively. RtAudioFormat userFormat; RtAudioFormat deviceFormat[2]; // Playback and record, respectively. StreamMutex mutex; CallbackInfo callbackInfo; ConvertInfo convertInfo[2]; double streamTime; // Number of elapsed seconds since the stream started. #if defined(HAVE_GETTIMEOFDAY) struct timeval lastTickTimestamp; #endif RtApiStream() :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } }; typedef S24 Int24; typedef signed short Int16; typedef signed int Int32; typedef float Float32; typedef double Float64; std::ostringstream errorStream_; std::string errorText_; bool showWarnings_; RtApiStream stream_; bool firstErrorOccurred_; /*! Protected, api-specific method that attempts to open a device with the given parameters. This function MUST be implemented by all subclasses. If an error is encountered during the probe, a "warning" message is reported and FAILURE is returned. A successful probe is indicated by a return value of SUCCESS. */ virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ); //! A protected function used to increment the stream time. void tickStreamTime( void ); //! Protected common method to clear an RtApiStream structure. void clearStreamInfo(); /*! Protected common method that throws an RtAudioError (type = INVALID_USE) if a stream is not open. */ void verifyStream( void ); //! Protected common error method to allow global control over error handling. void error( RtAudioError::Type type ); /*! Protected method used to perform format, channel number, and/or interleaving conversions between the user and device buffers. */ void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ); //! Protected common method used to perform byte-swapping on buffers. void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ); //! Protected common method that returns the number of bytes for a given format. unsigned int formatBytes( RtAudioFormat format ); //! Protected common method that sets up the parameters for buffer conversion. void setConvertInfo( StreamMode mode, unsigned int firstChannel ); }; // **************************************************************** // // // Inline RtAudio definitions. // // **************************************************************** // inline RtAudio::Api RtAudio :: getCurrentApi( void ) { return rtapi_->getCurrentApi(); } inline unsigned int RtAudio :: getDeviceCount( void ) { return rtapi_->getDeviceCount(); } inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } inline unsigned int RtAudio :: getDefaultInputDevice( void ) { return rtapi_->getDefaultInputDevice(); } inline unsigned int RtAudio :: getDefaultOutputDevice( void ) { return rtapi_->getDefaultOutputDevice(); } inline void RtAudio :: closeStream( void ) { return rtapi_->closeStream(); } inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } inline bool RtAudio :: isStreamOpen( void ) const { return rtapi_->isStreamOpen(); } inline bool RtAudio :: isStreamRunning( void ) const { return rtapi_->isStreamRunning(); } inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); } inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); } inline void RtAudio :: showWarnings( bool value ) { rtapi_->showWarnings( value ); } // RtApi Subclass prototypes. #if defined(__MACOSX_CORE__) #include class RtApiCore: public RtApi { public: RtApiCore(); ~RtApiCore(); RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); unsigned int getDefaultOutputDevice( void ); unsigned int getDefaultInputDevice( void ); void closeStream( void ); void startStream( void ); void stopStream( void ); void abortStream( void ); long getStreamLatency( void ); // This function is intended for internal use only. It must be // public because it is called by the internal callback handler, // which is not a member of RtAudio. External use of this function // will most likely produce highly undesireable results! bool callbackEvent( AudioDeviceID deviceId, const AudioBufferList *inBufferList, const AudioBufferList *outBufferList ); private: bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ); static const char* getErrorCode( OSStatus code ); }; #endif #if defined(__UNIX_JACK__) class RtApiJack: public RtApi { public: RtApiJack(); ~RtApiJack(); RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); void startStream( void ); void stopStream( void ); void abortStream( void ); long getStreamLatency( void ); // This function is intended for internal use only. It must be // public because it is called by the internal callback handler, // which is not a member of RtAudio. External use of this function // will most likely produce highly undesireable results! bool callbackEvent( unsigned long nframes ); private: bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ); bool shouldAutoconnect_; }; #endif #if defined(__WINDOWS_ASIO__) class RtApiAsio: public RtApi { public: RtApiAsio(); ~RtApiAsio(); RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); void startStream( void ); void stopStream( void ); void abortStream( void ); long getStreamLatency( void ); // This function is intended for internal use only. It must be // public because it is called by the internal callback handler, // which is not a member of RtAudio. External use of this function // will most likely produce highly undesireable results! bool callbackEvent( long bufferIndex ); private: std::vector devices_; void saveDeviceInfo( void ); bool coInitialized_; bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ); }; #endif #if defined(__WINDOWS_DS__) class RtApiDs: public RtApi { public: RtApiDs(); ~RtApiDs(); RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; } unsigned int getDeviceCount( void ); unsigned int getDefaultOutputDevice( void ); unsigned int getDefaultInputDevice( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); void startStream( void ); void stopStream( void ); void abortStream( void ); long getStreamLatency( void ); // This function is intended for internal use only. It must be // public because it is called by the internal callback handler, // which is not a member of RtAudio. External use of this function // will most likely produce highly undesireable results! void callbackEvent( void ); private: bool coInitialized_; bool buffersRolling; long duplexPrerollBytes; std::vector dsDevices; bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ); }; #endif #if defined(__WINDOWS_WASAPI__) struct IMMDeviceEnumerator; class RtApiWasapi : public RtApi { public: RtApiWasapi(); ~RtApiWasapi(); RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); unsigned int getDefaultOutputDevice( void ); unsigned int getDefaultInputDevice( void ); void closeStream( void ); void startStream( void ); void stopStream( void ); void abortStream( void ); private: bool coInitialized_; IMMDeviceEnumerator* deviceEnumerator_; bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int* bufferSize, RtAudio::StreamOptions* options ); static DWORD WINAPI runWasapiThread( void* wasapiPtr ); static DWORD WINAPI stopWasapiThread( void* wasapiPtr ); static DWORD WINAPI abortWasapiThread( void* wasapiPtr ); void wasapiThread(); }; #endif #if defined(__LINUX_ALSA__) class RtApiAlsa: public RtApi { public: RtApiAlsa(); ~RtApiAlsa(); RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); void startStream( void ); void stopStream( void ); void abortStream( void ); // This function is intended for internal use only. It must be // public because it is called by the internal callback handler, // which is not a member of RtAudio. External use of this function // will most likely produce highly undesireable results! void callbackEvent( void ); private: std::vector devices_; void saveDeviceInfo( void ); bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ); }; #endif #if defined(__LINUX_PULSE__) class RtApiPulse: public RtApi { public: ~RtApiPulse(); RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); void startStream( void ); void stopStream( void ); void abortStream( void ); // This function is intended for internal use only. It must be // public because it is called by the internal callback handler, // which is not a member of RtAudio. External use of this function // will most likely produce highly undesireable results! void callbackEvent( void ); private: std::vector devices_; void saveDeviceInfo( void ); bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ); }; #endif #if defined(__LINUX_OSS__) class RtApiOss: public RtApi { public: RtApiOss(); ~RtApiOss(); RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); void startStream( void ); void stopStream( void ); void abortStream( void ); // This function is intended for internal use only. It must be // public because it is called by the internal callback handler, // which is not a member of RtAudio. External use of this function // will most likely produce highly undesireable results! void callbackEvent( void ); private: bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ); }; #endif #if defined(__RTAUDIO_DUMMY__) class RtApiDummy: public RtApi { public: RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); } RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; } unsigned int getDeviceCount( void ) { return 0; } RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; } void closeStream( void ) {} void startStream( void ) {} void stopStream( void ) {} void abortStream( void ) {} private: bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, RtAudio::StreamOptions * /*options*/ ) { return false; } }; #endif #endif // Indentation settings for Vim and Emacs // // Local Variables: // c-basic-offset: 2 // indent-tabs-mode: nil // End: // // vim: et sts=2 sw=2 rtaudio-5.0.0/rtaudio.pc.in000644 000766 000024 00000000561 13151627174 015702 0ustar00garystaff000000 000000 prefix=@prefix@ exec_prefix=${prefix} libdir=${exec_prefix}/lib includedir=${prefix}/include/rtaudio Name: librtaudio Description: RtAudio - a set of C++ classes that provide a common API for realtime audio input/output Version: @PACKAGE_VERSION@ Requires: @req@ Libs: -L${libdir} -lrtaudio Libs.private: -lpthread Cflags: -pthread -I${includedir} @CPPFLAGS@ rtaudio-5.0.0/tests/000755 000766 000024 00000000000 13151630570 014434 5ustar00garystaff000000 000000 rtaudio-5.0.0/tests/audioprobe.cpp000644 000766 000024 00000006206 13151627174 017303 0ustar00garystaff000000 000000 /******************************************/ /* audioprobe.cpp by Gary P. Scavone, 2001 Probe audio system and prints device info. */ /******************************************/ #include "RtAudio.h" #include #include int main() { // Create an api map. std::map apiMap; apiMap[RtAudio::MACOSX_CORE] = "OS-X Core Audio"; apiMap[RtAudio::WINDOWS_ASIO] = "Windows ASIO"; apiMap[RtAudio::WINDOWS_DS] = "Windows Direct Sound"; apiMap[RtAudio::WINDOWS_WASAPI] = "Windows WASAPI"; apiMap[RtAudio::UNIX_JACK] = "Jack Client"; apiMap[RtAudio::LINUX_ALSA] = "Linux ALSA"; apiMap[RtAudio::LINUX_PULSE] = "Linux PulseAudio"; apiMap[RtAudio::LINUX_OSS] = "Linux OSS"; apiMap[RtAudio::RTAUDIO_DUMMY] = "RtAudio Dummy"; std::vector< RtAudio::Api > apis; RtAudio :: getCompiledApi( apis ); std::cout << "\nRtAudio Version " << RtAudio::getVersion() << std::endl; std::cout << "\nCompiled APIs:\n"; for ( unsigned int i=0; i #include #include /* typedef char MY_TYPE; #define FORMAT RTAUDIO_SINT8 */ typedef signed short MY_TYPE; #define FORMAT RTAUDIO_SINT16 /* typedef S24 MY_TYPE; #define FORMAT RTAUDIO_SINT24 typedef signed long MY_TYPE; #define FORMAT RTAUDIO_SINT32 typedef float MY_TYPE; #define FORMAT RTAUDIO_FLOAT32 typedef double MY_TYPE; #define FORMAT RTAUDIO_FLOAT64 */ void usage( void ) { // Error function in case of incorrect command-line // argument specifications std::cout << "\nuseage: duplex N fs \n"; std::cout << " where N = number of channels,\n"; std::cout << " fs = the sample rate,\n"; std::cout << " iDevice = optional input device to use (default = 0),\n"; std::cout << " oDevice = optional output device to use (default = 0),\n"; std::cout << " iChannelOffset = an optional input channel offset (default = 0),\n"; std::cout << " and oChannelOffset = optional output channel offset (default = 0).\n\n"; exit( 0 ); } int inout( void *outputBuffer, void *inputBuffer, unsigned int /*nBufferFrames*/, double /*streamTime*/, RtAudioStreamStatus status, void *data ) { // Since the number of input and output channels is equal, we can do // a simple buffer copy operation here. if ( status ) std::cout << "Stream over/underflow detected." << std::endl; unsigned int *bytes = (unsigned int *) data; memcpy( outputBuffer, inputBuffer, *bytes ); return 0; } int main( int argc, char *argv[] ) { unsigned int channels, fs, bufferBytes, oDevice = 0, iDevice = 0, iOffset = 0, oOffset = 0; // Minimal command-line checking if (argc < 3 || argc > 7 ) usage(); RtAudio adac; if ( adac.getDeviceCount() < 1 ) { std::cout << "\nNo audio devices found!\n"; exit( 1 ); } channels = (unsigned int) atoi(argv[1]); fs = (unsigned int) atoi(argv[2]); if ( argc > 3 ) iDevice = (unsigned int) atoi(argv[3]); if ( argc > 4 ) oDevice = (unsigned int) atoi(argv[4]); if ( argc > 5 ) iOffset = (unsigned int) atoi(argv[5]); if ( argc > 6 ) oOffset = (unsigned int) atoi(argv[6]); // Let RtAudio print messages to stderr. adac.showWarnings( true ); // Set the same number of channels for both input and output. unsigned int bufferFrames = 512; RtAudio::StreamParameters iParams, oParams; iParams.deviceId = iDevice; iParams.nChannels = channels; iParams.firstChannel = iOffset; oParams.deviceId = oDevice; oParams.nChannels = channels; oParams.firstChannel = oOffset; if ( iDevice == 0 ) iParams.deviceId = adac.getDefaultInputDevice(); if ( oDevice == 0 ) oParams.deviceId = adac.getDefaultOutputDevice(); RtAudio::StreamOptions options; //options.flags |= RTAUDIO_NONINTERLEAVED; bufferBytes = bufferFrames * channels * sizeof( MY_TYPE ); try { adac.openStream( &oParams, &iParams, FORMAT, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options ); } catch ( RtAudioError& e ) { std::cout << '\n' << e.getMessage() << '\n' << std::endl; exit( 1 ); } // Test RtAudio functionality for reporting latency. std::cout << "\nStream latency = " << adac.getStreamLatency() << " frames" << std::endl; try { adac.startStream(); char input; std::cout << "\nRunning ... press to quit (buffer frames = " << bufferFrames << ").\n"; std::cin.get(input); // Stop the stream. adac.stopStream(); } catch ( RtAudioError& e ) { std::cout << '\n' << e.getMessage() << '\n' << std::endl; goto cleanup; 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Scavone, 2007 Play a specified raw file. It is necessary that the file be of the same data format as defined below. */ /******************************************/ #include "RtAudio.h" #include #include #include #include /* typedef char MY_TYPE; #define FORMAT RTAUDIO_SINT8 #define SCALE 127.0 */ typedef signed short MY_TYPE; #define FORMAT RTAUDIO_SINT16 #define SCALE 32767.0 /* typedef S24 MY_TYPE; #define FORMAT RTAUDIO_SINT24 #define SCALE 8388607.0 typedef signed int MY_TYPE; #define FORMAT RTAUDIO_SINT32 #define SCALE 2147483647.0 typedef float MY_TYPE; #define FORMAT RTAUDIO_FLOAT32 #define SCALE 1.0; typedef double MY_TYPE; #define FORMAT RTAUDIO_FLOAT64 #define SCALE 1.0; */ // Platform-dependent sleep routines. #if defined( __WINDOWS_ASIO__ ) || defined( __WINDOWS_DS__ ) || defined( __WINDOWS_WASAPI__ ) #include #define SLEEP( milliseconds ) Sleep( (DWORD) milliseconds ) #else // Unix variants #include #define SLEEP( milliseconds ) usleep( (unsigned long) (milliseconds * 1000.0) ) #endif void usage( void ) { // Error function in case of incorrect command-line // argument specifications std::cout << "\nuseage: playraw N fs file \n"; std::cout << " where N = number of channels,\n"; std::cout << " fs = the sample rate, \n"; std::cout << " file = the raw file to play,\n"; std::cout << " device = optional device to use (default = 0),\n"; std::cout << " and channelOffset = an optional channel offset on the device (default = 0).\n\n"; exit( 0 ); } struct OutputData { FILE *fd; unsigned int channels; }; // Interleaved buffers int output( void *outputBuffer, void * /*inputBuffer*/, unsigned int nBufferFrames, double /*streamTime*/, RtAudioStreamStatus /*status*/, void *data ) { OutputData *oData = (OutputData*) data; // In general, it's not a good idea to do file input in the audio // callback function but I'm doing it here because I don't know the // length of the file we are reading. unsigned int count = fread( outputBuffer, oData->channels * sizeof( MY_TYPE ), nBufferFrames, oData->fd); if ( count < nBufferFrames ) { unsigned int bytes = (nBufferFrames - count) * oData->channels * sizeof( MY_TYPE ); unsigned int startByte = count * oData->channels * sizeof( MY_TYPE ); memset( (char *)(outputBuffer)+startByte, 0, bytes ); return 1; } return 0; } int main( int argc, char *argv[] ) { unsigned int channels, fs, bufferFrames, device = 0, offset = 0; char *file; // minimal command-line checking if ( argc < 4 || argc > 6 ) usage(); RtAudio dac; if ( dac.getDeviceCount() < 1 ) { std::cout << "\nNo audio devices found!\n"; exit( 0 ); } channels = (unsigned int) atoi( argv[1]) ; fs = (unsigned int) atoi( argv[2] ); file = argv[3]; if ( argc > 4 ) device = (unsigned int) atoi( argv[4] ); if ( argc > 5 ) offset = (unsigned int) atoi( argv[5] ); OutputData data; data.fd = fopen( file, "rb" ); if ( !data.fd ) { std::cout << "Unable to find or open file!\n"; exit( 1 ); } // Set our stream parameters for output only. bufferFrames = 512; RtAudio::StreamParameters oParams; oParams.deviceId = device; oParams.nChannels = channels; oParams.firstChannel = offset; if ( device == 0 ) oParams.deviceId = dac.getDefaultOutputDevice(); data.channels = channels; try { dac.openStream( &oParams, NULL, FORMAT, fs, &bufferFrames, &output, (void *)&data ); dac.startStream(); } catch ( RtAudioError& e ) { std::cout << '\n' << e.getMessage() << '\n' << std::endl; goto cleanup; } std::cout << "\nPlaying raw file " << file << " (buffer frames = " << bufferFrames << ")." << std::endl; while ( 1 ) { SLEEP( 100 ); // wake every 100 ms to check if we're done if ( dac.isStreamRunning() == false ) break; } cleanup: fclose( data.fd ); dac.closeStream(); return 0; } rtaudio-5.0.0/tests/playsaw.cpp000644 000766 000024 00000013475 13151627174 016640 0ustar00garystaff000000 000000 /******************************************/ /* playsaw.cpp by Gary P. Scavone, 2006 This program will output sawtooth waveforms of different frequencies on each channel. */ /******************************************/ #include "RtAudio.h" #include #include /* typedef char MY_TYPE; #define FORMAT RTAUDIO_SINT8 #define SCALE 127.0 */ typedef signed short MY_TYPE; #define FORMAT RTAUDIO_SINT16 #define SCALE 32767.0 /* typedef S24 MY_TYPE; #define FORMAT RTAUDIO_SINT24 #define SCALE 8388607.0 typedef signed long MY_TYPE; #define FORMAT RTAUDIO_SINT32 #define SCALE 2147483647.0 typedef float MY_TYPE; #define FORMAT RTAUDIO_FLOAT32 #define SCALE 1.0 typedef double MY_TYPE; #define FORMAT RTAUDIO_FLOAT64 #define SCALE 1.0 */ // Platform-dependent sleep routines. #if defined( __WINDOWS_ASIO__ ) || defined( __WINDOWS_DS__ ) || defined( __WINDOWS_WASAPI__ ) #include #define SLEEP( milliseconds ) Sleep( (DWORD) milliseconds ) #else // Unix variants #include #define SLEEP( milliseconds ) usleep( (unsigned long) (milliseconds * 1000.0) ) #endif #define BASE_RATE 0.005 #define TIME 1.0 void usage( void ) { // Error function in case of incorrect command-line // argument specifications std::cout << "\nuseage: playsaw N fs