-*.[oa]
+objects
foo*
bar*
para_*
confdefs.h
conftest
conftest.c
-
+GIT-VERSION-FILE
--- /dev/null
+
+ GNU LESSER GENERAL PUBLIC LICENSE
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+ <signature of Ty Coon>, 1 April 1990
+ Ty Coon, President of Vice
+
+That's all there is to it!
+
+
Gerd Becker <gerd.mac@gmx.net> (MacOs testing)
+Fabrice Bellard (FFmpeg)
+
Lorenzo Bettini <bettini@dsi.unifi.it> (gengetopt)
Ricardo Cerqueira <rmc@rccn.net> (mp3info)
Ian McDonald <imcdnzl@gmail.com> (dccp example code)
+Loren Merritt (FFmpeg)
+
Simon Morlat <simon.morlat@linphone.org> (ortp)
Christof Müller (bug reports)
M. Hari Nezumi <magenta@trikuare.cx> (AudioCompress)
+Michael Niedermayer <michaelni@gmx.at> (FFmpeg)
+
Manuel Odendahl <manuel-poc@bl0rg.net> (poc)
Guillaume Outters <guillaume.outters@free.fr> (mosx-mpg123)
Linus Torvalds <torvalds@osdl.org> (for giving us one hell of an
operating system [quote taken from README.linux for DOOM v1.666])
+
+Jean-Marc Valin (speex)
# defined locally in source files will be included in the documentation.
# If set to NO only classes defined in header files are included.
-EXTRACT_LOCAL_CLASSES = YES
+EXTRACT_LOCAL_CLASSES = NO
# This flag is only useful for Objective-C code. When set to YES local
# methods, which are defined in the implementation section but not in
HAVE_FAAD \
HAVE_OGGVORBIS \
__GNUC__=4 \
- __GNUC_MINOR__=4
+ __GNUC_MINOR__=4 \
+ HAVE_UCRED
# If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
# this tag can be used to specify a list of macro names that should be expanded.
* Runs on Linux, Mac OS, FreeBSD, NetBSD, Solaris and probably other
Unixes
- * Mp3, ogg vorbis, aac(m4a) support
+ * Mp3, ogg/vorbis, ogg/speex, aac (m4a) and wma support
* Local or remote http, dccp, and udp network audio streaming
* IPv6 support
* Forward error correction allows receivers to recover from packet losses
--- /dev/null
+#!/bin/sh
+
+if test $# -ne 1; then
+ echo >&2 "usage: $0 <version file>"
+ exit 1
+fi
+
+GVF="$1"
+DEF_VER="unnamed_version"
+
+LF='
+'
+
+# First see if there is a version file (included in release tarballs),
+# then try git-describe, then default.
+if test -f VERSION
+then
+ VN=$(cat VERSION) || VN="$DEF_VER"
+elif test -d .git -o -f .git &&
+ VN=$(git describe --abbrev=4 HEAD 2>/dev/null) &&
+ case "$VN" in
+ *$LF*) (exit 1) ;;
+ v[0-9]*)
+ git update-index -q --refresh
+ test -z "$(git diff-index --name-only HEAD --)" ||
+ VN="$VN-dirty" ;;
+ esac
+then
+ VN=$(echo "$VN" | sed -e 's/-/./g');
+else
+ VN="$DEF_VER"
+fi
+
+VN=$(expr "$VN" : v*'\(.*\)')
+
+if test -r $GVF
+then
+ VC=$(sed -e 's/^GIT_VERSION = //' <$GVF)
+else
+ VC=unset
+fi
+test "$VN" = "$VC" || {
+ echo >&2 "GIT_VERSION = $VN"
+ echo "GIT_VERSION = $VN" >$GVF
+}
-INSTALL
-=======
-
-----
Any knowledge of how to work with mouse and icons is not required.
----------------------------
-Install all needed packages
----------------------------
-See
-<<
-<a href="REQUIREMENTS.html"> REQUIREMENTS </a>
->>
-for a list of required software. You don't need everything listed
-there. In particular, mp3, ogg vorbis and aac support are all
-optional. The configure script will detect what is installed on your
-system and will only try to build those executables that can be built
-with your setup.
-
-Note that no special library (not even the mp3 decoding library libmad)
-is needed for para_server if you only want to stream mp3 files.
-Also, it's fine to use para_server on a box without sound card as
-para_server only sends the audio stream to connected clients.
-
--------------------------
-Install server and client
--------------------------
-
-Install the package on all machines, you'd like this software to run on:
-
- (./configure && make) > /dev/null
-
-There should be no errors but probably some warnings about missing
-software packages which usually implies that not all audio formats will
-be supported. If headers or libs are installed at unusual locations
-you might need to tell the configure script where to find them. Try
-
- ./configure --help
-
-to see a list of options. If the paraslash package was compiled
-successfully, execute as root,
-
- make install
-
------------------------------------
-Setup user list and create rsa keys
------------------------------------
-
-If you already have your rsa keys, skip this step. If you are new
-to paraslash, you have to generate an rsa key pair for each user you
-want to allow to connect. You need at least one user.
-
-Let's assume that you'd like to run the server on host server_host
-as user foo, and that you want to connect from client_host as user bar.
-
-As foo@server_host, create ~/.paraslash/server.users by typing the
-following commands:
-
- user=bar
- target=~/.paraslash/server.users
- key=~/.paraslash/key.pub.$user
- perms=AFS_READ,AFS_WRITE,VSS_READ,VSS_WRITE
- mkdir -p ~/.paraslash
- echo "user $user $key $perms" >> $target
-
-This gives "bar" the full privileges.
-
-Change to the "bar" account on client_host and generate the key-pair
-with the commands
-
- key=~/.paraslash/key.$LOGNAME
- mkdir -p ~/.paraslash
- (umask 077 && openssl genrsa -out $key)
-
-Next, extract its public part:
-
- pubkey=~/.paraslash/key.pub.$LOGNAME
- openssl rsa -in $key -pubout -out $pubkey
-
-and copy the public key just created to server_host (you may
-skip this step for a single-user setup, i.e. if foo=bar and
-server_host=client_host):
-
- scp $pubkey foo@server_host:.paraslash/
-
-Finally, tell para_client to connect to server_host:
-
- conf=~/.paraslash/client.conf
- echo 'hostname server_host' > $conf
-
------------------
-Start para_server
------------------
-
-Before starting the server make sure you have write permissions to
-the directory /var/paraslash.
-
- sudo chown $LOGNAME /var/paraslash
-
-Alternatively, use the --afs_socket Option to specify a different
-location for the afs command socket.
-
-For this first try, we'll use the info loglevel to make the output
-of para_server more verbose.
-
- para_server -l info
-
-Now you can use para_client to connect to the server and issue
-commands. Open a new shell (as "bar" on "client_host" in the above
-example) and try
-
- para_client help
- para_client si
-
-to retrieve the list of available commands and some server info.
-Don't proceed if this doesn't work.
-
--------------------
-Create the database
--------------------
-
- para_client init
-
-This creates some empty tables under ~/.paraslash/afs_database.
-You normally don't need to look at these tables, but it's good
-to know that you can start from scratch with
-
- rm -rf ~/.paraslash/afs_database
-
-in case something went wrong.
-
-Next, you need to fill the audio file table of that database with
-contents so that para_server knows about your audio files. Choose an
-absolute path to a directory containing some audio files and add them
-to the audio file table:
-
- para_client add /my/mp3/dir
-
-This might take a while, so it is a good idea to start with a directory
-containing not too many audio files. Note that the table only contains
-data about the audio files found, not the files themselves.
-
-Print a list of all audio files found with
-
- para_client ls
-
-------------------------
-Start streaming manually
-------------------------
-
- para_client play
- para_client stat 2
-
-This starts streaming and dumps some information about the current
-audio file to stdout.
-
-You should now be able to receive the stream and listen to it. If
-you have mpg123 or xmms handy, execute on client_host
-
- mpg123 http://server_host:8000/
-or
- xmms http://server_host:8000/
-
-Paraslash comes with its own receiving and playing software, which
-will be described next. Try the following on client_host (assuming
-Linux/ALSA and an mp3 stream):
-
- para_recv -l info -r 'http -i server_host' > file.mp3
- # (interrupt with CTRL+C after a few seconds)
- ls -l file.mp3 # should not be empty
- para_filter -f mp3dec -f wav < file.mp3 > file.wav
- ls -l file.wav # should be much bigger than file.mp3
- para_write -w alsa < file.wav
-
-If this works, proceed. Otherwise double check what is logged by
-para_server and use the --loglevel option of para_recv, para_filter
-and para_write to increase verbosity.
-
-Next, put the pieces together:
-
- para_recv -r 'http -i server_host' \
- | para_filter -f mp3dec -f wav \
- | para_write -w alsa
-
----------------------
-Configure para_audiod
----------------------
-
-In order to automatically start the right decoder at the right time
-and to offer to the clients some information on the current audio
-stream and on paraslash's internal state, you should run the local
-audio daemon, para_audiod, on every machine in your network which is
-supposed to play the audio stream. Try
-
- para_audiod -h
-
-for help. Usually you have to specify only server_host as the receiver
-specifier for each supported audio format, like this:
-
- para_audiod -l info -r 'mp3:http -i server_host'
-
-The preferred way to use para_audiod is to run it once at system start
-as an unprivileged user. para_audiod needs to create a "well-known"
-socket for the clients to connect to. The default path for this
-socket is
-
- /var/paraslash/audiod_socket.$HOSTNAME
-
-so the /var/paraslash directory should be writable for the user who
-runs para_audiod.
-
-If you want to change the location of the socket, use the --socket
-option for para_audiod or the config file ~/.paraslash/audiod.conf
-to change the default. Note that in this case you'll also have to
-specify the same value for para_audioc's --socket option.
-
-If para_server is playing, you should be able to listen to the audio
-stream as soon as para_audiod is started. Once it is running, try
-
- para_audioc stat
-
-That should dump some information to stdout. Other commands include
+From tarball:
- para_audioc off
- para_audioc on
- para_audioc sb
- para_audioc term
- para_audioc cycle
+ ./configure && make && sudo make install
---------------
-Start para_gui
---------------
+From git:
-para_gui reads the output of "para_audioc stat" and displays that
-information in a curses window. It also allows you to bind keys to
-arbitrary commands. There are several flavours of key-bindings:
+ ./autogen.sh && sudo make install
- - internal: These are the built-in commands that can not be
- changed (help, quit, loglevel, version...).
- - external: Shutdown curses before launching the given command.
- Useful for starting other ncurses programs from within
- para_gui, e.g. aumix or dialog scripts. Or, use the mbox
- output format to write a mailbox containing one mail for each
- (admissible) file the audio file selector knows about. Then
- start mutt from within para_gui to browse your collection!
- - display: Launch the command and display its stdout in
- para_gui's bottom window.
- - para: Like display, but start "para_client <specified
- command>" instead of "<specified command>".
+For details see the user manual:
-This concludes the installation notes. Next thing you might to have a look
-at is how to use paraslash's audio file selector. See
-<<
-<a href="README.afs.html"> README.afs</a>
->>
+ http://paraslash.systemlinux.org/manual.html
PACKAGE_VERSION := @PACKAGE_VERSION@
PACKAGE_STRING := @PACKAGE_STRING@
install_sh := @install_sh@
+cmdline_dir := @cmdline_dir@
build_date := $(shell date)
uname_s := $(shell uname -s 2>/dev/null || echo "UNKNOWN_OS")
uname_rs := $(shell uname -rs)
cc_version := $(shell $(CC) --version | head -n 1)
-codename := imaginary radiation
+codename := deterministic entropy
DEBUG_CPPFLAGS += -Wno-sign-compare -g -Wunused -Wundef -W
DEBUG_CPPFLAGS += -Wredundant-decls
CPPFLAGS += @SSL_CPPFLAGS@
CPPFLAGS += @ncurses_cppflags@
CPPFLAGS += @arch_cppflags@
+CPPFLAGS += -I/usr/local/include
+CPPFLAGS += -I$(cmdline_dir)
+CPPFLAGS += @osl_cppflags@
+CPPFLAGS += -DGIT_VERSION='"$(GIT_VERSION)"'
-BINARIES = para_server para_client para_audioc para_recv \
- para_filter para_write para_fsck para_afh @extra_binaries@
-man_binaries := $(BINARIES)
-man_pages := $(patsubst %, man/man1/%.1, $(man_binaries))
-man_pages_in := $(patsubst %, web/%.man.in.html, $(man_binaries))
+man_pages := $(patsubst %, man/man1/%.1, @executables@)
+man_pages_in := $(patsubst %, web/%.man.in.html, @executables@)
ggo_dir := ggo
+object_dir := objects
+man_dir := man/man1
-m4_ggos := afh audioc audiod client filter fsck gui recv server write
-all_ggos := $(m4_ggos) dccp_recv oggdec_filter alsa_write oss_write fade http_recv \
+m4_ggos := afh audioc audiod client filter gui recv server write
+all_ggos := $(m4_ggos) dccp_recv alsa_write oss_write fade http_recv \
osx_write udp_recv amp_filter compress_filter file_write \
- grab_client mp3dec_filter
-ggo_generated := $(addsuffix .cmdline.c, $(all_ggos)) $(addsuffix .cmdline.h, $(all_ggos)) \
- $(addsuffix .ggo, $(addprefix $(ggo_dir)/,$(m4_ggos)))
+ mp3dec_filter prebuffer_filter
+ggo_generated := $(addsuffix .ggo, $(addprefix $(ggo_dir)/,$(m4_ggos)))
+cmdline_generated := $(addprefix $(cmdline_dir)/,$(addsuffix .cmdline.c, $(all_ggos)) \
+ $(addsuffix .cmdline.h, $(all_ggos)))
autocrap := config.h.in configure
tarball_pfx := @PACKAGE_TARNAME@-$(PACKAGE_VERSION)
-tarball_delete = web versions pics .changelog_before_cvs .changelog_cvs .gitignore
-tarball_delete := $(patsubst %,$(tarball_pfx)/%,$(tarball_delete))
-tarball_add := $(ggo_generated) $(autocrap)
+tarball_delete := $(addprefix $(tarball_pfx)/,\
+ web versions .changelog_before_cvs .changelog_cvs .gitignore\
+ $(ggo_dir) skencil)
tarball := @PACKAGE_TARNAME@-$(PACKAGE_VERSION).tar.bz2
-.PHONY: clean distclean maintainer-clean install man tarball
-all: $(BINARIES) $(man_pages)
+# To put more focus on warnings, be less verbose as default
+# Use 'make V=1' to see the full commands
+ifdef V
+ ifeq ("$(origin V)", "command line")
+ BUILD_VERBOSE = $(V)
+ endif
+endif
+ifndef BUILD_VERBOSE
+ BUILD_VERBOSE = 0
+endif
+ifeq ($(BUILD_VERBOSE),1)
+ Q =
+else
+ Q = @
+endif
+
+.PHONY: all clean distclean maintainer-clean install man tarball\
+ .FORCE-GIT-VERSION-FILE
+all: @executables@ $(man_pages)
man: $(man_pages)
tarball: $(tarball)
-*.o: para.h config.h gcc-compat.h
+GIT-VERSION-FILE: .FORCE-GIT-VERSION-FILE
+ @./GIT-VERSION-GEN GIT-VERSION-FILE
+-include GIT-VERSION-FILE
+@executables@: GIT-VERSION-FILE
-include Makefile.deps
-include $(ggo_dir)/makefile
-
-%_command_list.c %_command_list.h: %.cmd
- ./command_util.sh c < $< >$@
- ./command_util.sh h < $< >$(@:%.c=%.h)
+-include $(ggo_dir)/makefile
+%_command_list.c: %.cmd
+ @[ -z "$(Q)" ] || echo 'GEN $@'
+ $(Q) ./command_util.sh c < $< >$@
+%_command_list.h: %.cmd
+ @[ -z "$(Q)" ] || echo 'GEN $@'
+ $(Q) ./command_util.sh h < $< >$@
%_command_list.man: %.cmd
- ./command_util.sh man < $< > $@
-
-server_command_lists = server_command_list.man afs_command_list.man
-man/man1/para_server.1: para_server $(server_command_lists)
- mkdir -p man/man1
- opts="-h --detailed-help -N `for i in $(server_command_lists); do printf "%s\n" "-i $$i"; done`"; \
+ @[ -z "$(Q)" ] || echo 'GEN $@'
+ $(Q) ./command_util.sh man < $< > $@
+
+server_command_lists_ch = server_command_list.c afs_command_list.c \
+ server_command_list.h afs_command_list.h
+server_command_lists_man = server_command_list.man afs_command_list.man
+man/man1/para_server.1: para_server $(server_command_lists_man) | $(man_dir)
+ @[ -z "$(Q)" ] || echo 'HELP2MAN $<'
+ $(Q) opts="-h --detailed-help -N `for i in $(server_command_lists_man); do printf "%s\n" "-i $$i"; done`"; \
help2man $$opts ./para_server > $@
-man/man1/para_audiod.1: para_audiod audiod_command_list.man
- mkdir -p man/man1
- help2man -h --detailed-help -N -i audiod_command_list.man ./para_audiod > $@
+man/man1/para_audiod.1: para_audiod audiod_command_list.man | $(man_dir)
+ @[ -z "$(Q)" ] || echo 'HELP2MAN $<'
+ $(Q) help2man -h --detailed-help -N -i audiod_command_list.man ./para_audiod > $@
-man/man1/%.1: %
- mkdir -p man/man1
- help2man -h --detailed-help -N ./$< > $@
+man/man1/%.1: % | $(man_dir)
+ @[ -z "$(Q)" ] || echo 'HELP2MAN $<'
+ $(Q) help2man -h --detailed-help -N ./$< > $@
man/html/%.html: man/man1/%.1
- mkdir -p man/html
- man2html $< > $@
+ @[ -z "$(Q)" ] || echo 'MAN2HTML $<'
+ $(Q) mkdir -p man/html
+ $(Q) man2html $< > $@
web/%.man.in.html: man/man1/%.1
- man2html $< | sed -e '/^<\/BODY>/,$$d' -e '1,/<\/HEAD><BODY>/d' > $@
-
-
-oggdec_filter.o: oggdec_filter.c
- $(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @oggvorbis_cppflags@ $<
-ogg_afh.o: ogg_afh.c
- $(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @oggvorbis_cppflags@ $<
-
-mp3dec_filter.o: mp3dec_filter.c
- $(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @mad_cppflags@ $<
-
-aacdec_filter.o: aacdec_filter.c
- $(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @faad_cppflags@ $<
-
-aac_common.o: aac_common.c
- $(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @faad_cppflags@ $<
-
-aac_afh.o: aac_afh.c
- $(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @faad_cppflags@ $<
-
-%.cmdline.o: %.cmdline.c
- $(CC) -c $(CPPFLAGS) $<
+ @[ -z "$(Q)" ] || echo 'MAN2HTML $<'
+ $(Q) mkdir -p man/html
+ $(Q) man2html $< | sed -e '/^<\/BODY>/,$$d' -e '1,/<\/HEAD><BODY>/d' > $@
+
+$(object_dir):
+ mkdir -p $@
+$(man_dir):
+ mkdir -p $@
+
+$(object_dir)/spx_common.o: spx_common.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @ogg_cppflags@ $<
+
+$(object_dir)/spxdec_filter.o: spxdec_filter.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @ogg_cppflags@ $<
+
+$(object_dir)/spx_afh.o: spx_afh.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @ogg_cppflags@ $<
+
+$(object_dir)/oggdec_filter.o: oggdec_filter.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @ogg_cppflags@ $<
+
+$(object_dir)/ogg_afh.o: ogg_afh.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @ogg_cppflags@ $<
+
+$(object_dir)/ogg_afh_common.o: ogg_afh_common.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @ogg_cppflags@ $<
+
+$(object_dir)/mp3dec_filter.o: mp3dec_filter.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @mad_cppflags@ $<
+
+$(object_dir)/aacdec_filter.o: aacdec_filter.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @faad_cppflags@ $<
+
+$(object_dir)/aac_common.o: aac_common.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @faad_cppflags@ $<
+
+$(object_dir)/aac_afh.o: aac_afh.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) @faad_cppflags@ $<
+
+$(object_dir)/%.cmdline.o: $(cmdline_dir)/%.cmdline.c $(cmdline_dir)/%.cmdline.h | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c $(CPPFLAGS) -Wno-unused-function -o $@ $<
+
+$(object_dir)/%.o: %.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'CC $<'
+ $(Q) $(CC) -c -o $@ $(CPPFLAGS) $(DEBUG_CPPFLAGS) $<
+
+$(object_dir)/%.cmdline.d: $(cmdline_dir)/%.cmdline.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'DEP $<'
+ $(Q) ./depend.sh $(object_dir) $(cmdline_dir) $(CPPFLAGS) $< > $@
+
+$(object_dir)/%.d: %.c | $(object_dir)
+ @[ -z "$(Q)" ] || echo 'DEP $<'
+ $(Q) ./depend.sh $(object_dir) $(cmdline_dir) $(CPPFLAGS) $< > $@
+
+recv_objs := $(addprefix $(object_dir)/, @recv_objs@)
+filter_objs := $(addprefix $(object_dir)/, @filter_objs@)
+client_objs := $(addprefix $(object_dir)/, @client_objs@)
+gui_objs := $(addprefix $(object_dir)/, @gui_objs@)
+audiod_objs := $(addprefix $(object_dir)/, @audiod_objs@)
+audioc_objs := $(addprefix $(object_dir)/, @audioc_objs@)
+fade_objs := $(addprefix $(object_dir)/, @fade_objs@)
+server_objs := $(addprefix $(object_dir)/, @server_objs@)
+write_objs := $(addprefix $(object_dir)/, @write_objs@)
+afh_objs := $(addprefix $(object_dir)/, @afh_objs@)
+
+all_objs := $(recv_objs) $(filter_objs) $(client_objs) $(gui_objs) \
+ $(audiod_objs) $(audioc_objs) $(fade_objs) $(server_objs) \
+ $(write_objs) $(afh_objs)
+
+ifeq ($(findstring clean, $(MAKECMDGOALS)),)
+-include $(all_objs:.o=.d)
+endif
-%.o: %.c
- $(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) $<
+para_recv: $(recv_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) $(recv_objs) -o $@ @recv_ldflags@
-para_recv: @recv_objs@
- $(CC) $(LDFLAGS) @recv_objs@ -o $@ @recv_ldflags@
+para_filter: $(filter_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) $(filter_objs) -o $@ @filter_ldflags@
-para_filter: @filter_objs@
- $(CC) $(LDFLAGS) @filter_objs@ -o $@ @filter_ldflags@
+para_client: $(client_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) -o $@ $(client_objs) @client_ldflags@
-para_client: @client_objs@
- $(CC) $(LDFLAGS) -o $@ @client_objs@ @client_ldflags@
+para_gui: $(gui_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) -o $@ $(gui_objs) -lncurses
-para_gui: @gui_objs@
- $(CC) $(LDFLAGS) -o $@ @gui_objs@ -lncurses
+para_audiod: audiod_command_list.c audiod_command_list.h $(audiod_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) -o $@ $(audiod_objs) @audiod_ldflags@
-para_audiod: @audiod_objs@
- $(CC) $(LDFLAGS) -o $@ @audiod_objs@ @audiod_ldflags@
+para_audioc: $(audioc_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) -o $@ $(audioc_objs) @audioc_ldflags@
-para_audioc: @audioc_objs@
- $(CC) $(LDFLAGS) -o $@ @audioc_objs@ @audioc_ldflags@
+para_fade: $(fade_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) -o $@ $(fade_objs) @fade_ldflags@
-para_fade: @fade_objs@
- $(CC) $(LDFLAGS) -o $@ @fade_objs@ @fade_ldflags@
+para_server: $(server_command_lists_ch) $(server_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) -o $@ $(server_objs) @server_ldflags@
-para_server: @server_objs@
- $(CC) $(LDFLAGS) -o $@ @server_objs@ @server_ldflags@
+para_write: $(write_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) -o $@ $(write_objs) @write_ldflags@
-para_fsck: @fsck_objs@
- $(CC) $(LDFLAGS) -o $@ @fsck_objs@ @fsck_ldflags@
+para_afh: $(afh_objs)
+ @[ -z "$(Q)" ] || echo 'LD $@'
+ $(Q) $(CC) $(LDFLAGS) -o $@ $(afh_objs) @afh_ldflags@
-para_write: @write_objs@
- $(CC) $(LDFLAGS) -o $@ @write_objs@ @write_ldflags@
+clean:
+ @[ -z "$(Q)" ] || echo 'CLEAN'
+ $(Q) rm -f @executables@ $(object_dir)/*.o
-para_afh: @afh_objs@
- $(CC) $(LDFLAGS) -o $@ @afh_objs@ @afh_ldflags@
+clean2: clean
+ @[ -z "$(Q)" ] || echo 'CLEAN2'
+ $(Q) rm -rf man $(object_dir)
+ $(Q) rm -f *_command_list.*
-clean:
- rm -f *.o $(BINARIES)
- rm -rf man
-distclean: clean
- rm -f Makefile autoscan.log config.status config.log && \
- rm -rf web/sync/* autom4te.cache aclocal.m4
- rm -f GPATH GRTAGS GSYMS GTAGS
+distclean: clean2
+ @[ -z "$(Q)" ] || echo 'DISTCLEAN'
+ $(Q) rm -f Makefile autoscan.log config.status config.log
+ $(Q) rm -rf autom4te.cache aclocal.m4
+ $(Q) rm -f GPATH GRTAGS GSYMS GTAGS
maintainer-clean: distclean
rm -f $(ggo_generated) *.tar.bz2 \
config.h configure \
config.h.in skencil/*.pdf skencil/*.ps
- rm -f *_command_list.* *.man man/man1/*
- rm -rf web_sync
+ rm -rf web_sync $(cmdline_dir)
install: all man
mkdir -p $(BINDIR) $(MANDIR)
- $(install_sh) -s -m 755 $(BINARIES) $(BINDIR)
+ $(install_sh) -s -m 755 @executables@ $(BINDIR)
$(install_sh) -m 644 $(man_pages) $(MANDIR)
mkdir -p $(VARDIR) >/dev/null 2>&1 || true # not fatal, so don't complain
-@PACKAGE_TARNAME@-$(PACKAGE_VERSION).tar.bz2: $(tarball_add)
+$(tarball): $(cmdline_generated)
rm -rf $(tarball_pfx).tar.bz2 $(tarball_pfx)
git archive --format=tar --prefix=$(tarball_pfx)/ HEAD \
| tar --delete $(tarball_delete) > $(tarball_pfx).tar
- mkdir $(tarball_pfx)
- cp -r $(tarball_add) $(tarball_pfx)
+ mkdir -p $(tarball_pfx)/$(cmdline_dir)
+ echo $(GIT_VERSION) > $(tarball_pfx)/VERSION
+ cp -r $(autocrap) $(tarball_pfx)
+ cp -r $(cmdline_generated) $(tarball_pfx)/$(cmdline_dir)
tar rf $(tarball_pfx).tar $(tarball_pfx)/*
rm -rf $(tarball_pfx)
bzip2 -9 $(tarball_pfx).tar
-NEWS
-====
+-----------------------------------------------
+0.4.6 (to be announced) "deterministic entropy"
+-----------------------------------------------
+
+ - For DCCP/OGG streams the audio file header is only sent once
+ at the beginning of the stream rather than periodically
+ every five seconds. This reduces network traffic and the
+ FEC group size.
+ - aacdec error message cleanups
+ - simplified color error handling
+
+--------------------------------------------
+0.4.5 (2010-12-17) "symmetric randomization"
+--------------------------------------------
+
+Bug fixes, internal cleanups and variable-sized FEC slices.
+
+ - Contains a fix for an invalid-free-bug in the ogg audio format
+ handler code.
+ - Switching off the DCCP sender works again.
+ - para_audiod handles crashes of para_server more robustly.
+ - Internal scheduler and writer cleanups.
+ - Reduced latency due to variable-sized FEC slices.
+ - Improved documentation and error diagnostics.
+ - The build of para_server is now optional, allowing the build
+ to succeed in case libosl is not installed.
+
+------------------------------------------
+0.4.4 (2010-08-06) "persistent regularity"
+------------------------------------------
+
+Support for yet another audio format, para_write improvements and
+bug fixes.
+
+ - Support for the speex codec.
+ - Support for sample formats other than 16 bit little endian.
+ - error2.h is now created by a perl script which speeds up configure
+ considerably.
+ - Fix a bug in the aac decoder which could lead to segfaults in
+ para_filter/para_audiod.
+ - Fixes for autoconf-2.66.
+
+----------------------------------------
+0.4.3 (2010-07-05) "imaginary radiation"
+----------------------------------------
+
+Many improvements for the DCCP and the UDP transport, the new user
+manual and the usual mix of bug fixes and internal improvements.
+
+ - FEC support for the DCCP sender (Gerrit Renker). The new
+ --dccp_max_slice_size, --dccp_data_slices_per_group and
+ --dccp_slices_per_group options can be used to set the FEC
+ parameters for the DCCP transport.
+ - DNS lookups for UDP targets (Gerrit Renker).
+ - The new user manual replaces the README, README.afs, REQUIREMENTS
+ and INSTALL documents.
+ - Fix an end-of-file detection bug in the oggdec filter.
+ - The new nonblock API.
+ - Both options of the oggdec filter have been removed.
+ - New debug mode for the internal scheduler.
+
+------------------------------------------
+0.4.2 (2010-04-23) "associative expansion"
+------------------------------------------
+
+It's been some time since the last release, but finally here is
+paraslash-0.4.2. The bulk of the changes comes from the new buffer
+tree API, but there are changes all over the tree. Mainly performance
+and usability improvements, but also quite some bug fixes.
+
+ - The new buffer tree API.
+ - DCCP: Support for CCID negotiation (Gerrit Renker).
+ - UDP robustness fixes.
+ - The --bufsize option for mp3dec is gone as it no longer makes sense
+ for the new buffer tree API.
+ - Fix audible buffer underruns for wma streams.
+ - The alsa writer no longer prints meaningless underrun durations.
+ - audiod: Defaults work also for udp streams. If no filter is
+ given for an audio format that is received via upd, fecdec is
+ automatically added as the first filter (along with the decoder).
+
+---------------------------------------
+0.4.1 (2009-12-22) "concurrent horizon"
+---------------------------------------
+
+Support for another audio format, minor feature enhancements and lots of bug
+fixes. All fixes that have been accumulated in the maint branch (in particular
+those mentionened in the 0.3.6 release notes) appear in this release as well.
+
+ - wma support.
+ - new afh option: --human to activate human-readable output.
+ - new server/audiod option: --log-timing to print timing information.
+ - build system improvements.
+ - source code documentation updates.
-------------------------------------
0.3.6 (2009-12-07) "cubic continuity"
- http/dccp: Do not send the audio file header twice.
- FEC: Timing improvements.
+----------------------------------------------
+0.4.0 (2009-11-10) "simultaneous independence"
+----------------------------------------------
+
+Two significant changes which require the new version number: The
+improved authentication dialog and the fact that the database code
+has been moved to a library, libosl. To use the new version, you have
+to generate new RSA keys, see INSTALL for details. A shell script is
+provided for conversion of the 0.3 database to the new 0.4 format.
+
+ - stronger crypto for client authentication
+ - the database code has been moved to a library
+ - improved status item handling
+ - cleanup of the build system
+ - The "-V" option now also prints the git version
+ - the new parser-friendly listing mode for the ls and stat commands
+ - mandatory rc4 encryption
+ - major audio format handler cleanups
+ - (id3,...) tags are no longer stored as a combined string in the database
+ - new mood methods: artist_matches, title_matches, comment_matches,
+ album_matches, year_maches, year.
+
--------------------------------------------
0.3.5 (2009-09-21) "symplectic separability"
--------------------------------------------
_Play, archive, rate and stream large audio sets happily_
-It contains the following programs:
+The paraslash package contains server and client software for
+network streaming as well as stand-alone utilities for decoding
+mp3, ogg, aac and wma files. See the user manual for details.
------------
-para_server
------------
-
-para_server streams binary audio data (mp3/oggvorbis/m4a files)
-over local and/or remote networks. It listens on a tcp port and
-accepts commands such as play, stop, pause, next from authenticated
-clients. However, there are many more commands.
-
-It supports three built-in network streaming methods (senders): http, dccp,
-or udp.
-
- * The http sender is recommended for public streams that can be played
- by any player like mpg123, xmms, itunes, winamp...
-
- * The dccp sender requires kernel support for the datagram congestion
- control protocol.
-
- * The udp sender is recommended for multicast LAN streaming.
-
-It is possible to activate more than one sender simultaneously.
-
-The built-in audio file selector of paraslash is used to manage your
-audio files. It maintains statistics on the usage of all available audio
-files such as last played time, and the number of times each file was
-selected.
-
-Its features include
-
- * attributes. Allow fine-grained audio file selection.
-
- * image table. For storing e.g. album cover art.
-
- * lyrics table. For storing lyrics.
-
- * playlist table. Stores arbitrary many playlists for later use.
-
- * mood mode. Audio file selection works by specifying mood
- methods involving attributes, pattern matching for file names
- and more. This allows rather sophisticated configurations
- and is explained in more detail in
-<<
-<a href="README.afs.html"> README.afs </a>
->>
-
- * rename detection. If files are moved or renamed, afs will
- recognize them despite of this change.
-
-Despite of all these features, paraslash is lightweight. The
-stripped binary of para_server with all its features compiled in
-mp3/ogg/aac support, http/dccp/udp support) is about 160K on i386
-under Linux. para_audiod (see below) is even smaller.
-
------------
-para_client
------------
-
-The client program to connect to para_server. paraslash commands
-are sent to para_server and the response is dumped to stdout. This
-can be used by any scripting language to produce user interfaces with
-little programming effort.
-
-All connections between para_server and para_client are encrypted by
-default. For each user of paraslash you must create a public/secret
-RSA key pair for authentication. The authenticated connection is
-encrypted with a symmetric rc4 session key.
-
----------
-para_recv
----------
-
-A command line http/dccp/udp stream grabber. The http mode of this
-tool can be used to receive data from any http streaming source.
-
------------
-para_filter
------------
-
-A filter program that converts from stdin and writes to stdout.
-
-para_filter combines several decoders (mp3, oggvorbis, aac) and a
-volume normalizer. New filters can be added easily. It is possible
-to "chain" any number of filters, like UNIX pipes.
-
-para_filter does not depend on other parts of paraslash, so it can
-be used as a stand-alone command line tool for audio decoding and
-volume normalization.
-
---------
-para_afh
---------
-
-A small stand-alone program that prints tech info about the given
-audio file to stdout. It can be instructed to print a "chunk table",
-an array of offsets within the audio file or to write the content of
-the audio file in complete chunks 'just in time'.
-
-This allows third-party streaming software that is unaware of
-the particular audio format to send complete frames in real
-time. Currently, mp3, ogg vorbis and aac are supported.
-
-----------
-para_write
-----------
-
-A modular audio stream writer. It supports a simple file writer
-output plug-in and optional wav/raw players for ALSA (Linux) and for
-coreaudio (Mac OS). para_write can also be used as a stand-alone wav
-or raw audio player.
-
------------
-para_audiod
------------
-
-The local daemon that collects information from para_server.
-
-It runs on the client side and connects to para_server. As soon as
-para_server announces the availability of an audio stream, para_audiod
-starts an appropriate receiver, any number of filters and a paraslash
-writer to play the stream. It is possible to capture the stream at
-any position in the filter chain.
-
-Moreover, para_audiod listens on a local socket and sends status
-information about para_server and para_audiod to local clients on
-request. Access via this local socket may be restricted by using Unix
-socket credentials, if available.
-
------------
-para_audioc
------------
-
-The client program which talks to para_audiod. Used to control
-para_audiod, to receive status info, or to grab the stream at any
-point in the filter chain.
-
-para_audioc (hence para_audiod) is needed by para_gui see below.
-
---------
-para_gui
---------
-
-Themable ncurses-based gui. It calls para_audioc and presents
-the obtained information in an ncurses window. para_gui provides
-key-bindings for the most common commands and new key-bindings can
-be added easily.
-
----------
-para_fade
----------
-
-A (oss-only) alarm clock and volume-fader.
-
----------------
-bash_completion
----------------
-
-A small bash script for inclusion in ~/.bashrc. It gives you command
-line completion for some paraslash commands.
-------
LICENSE
+++ /dev/null
-The audio file selector
-=======================
-
-Paraslash comes with a sophisticated audio file selector called *afs*.
-In the
-<<
-<a href="INSTALL.html">installation notes</a>,
->>
-only the "dummy" mode of afs was used which gets activated automatically if
-nothing else was specified. In this section the various features of afs are
-described.
-
-----------
-Attributes
-~~~~~~~~~~
-
-An attribute is simply a bit which can be set for each audio
-file individually. Up to 64 different attributes may be
-defined. For example, "pop", "rock", "blues", "jazz", "instrumental",
-"german_lyrics", "speech", whatever. It's up to you how many attributes
-you define and how you call them.
-
-A new attribute "test" is created by
-
- para_client addatt test
-and
- para_client lsatt
-
-lists all available attributes. You can set the "test" attribute for
-an audio file by executing
-
- para_client setatt test+ /path/to/the/audio/file
-
-Similarly, the "test" bit can be removed from an audio file with
-
- para_client setatt test- /path/to/the/audio/file
-
-Instead of a path you may use a shell wildcard pattern. The attribute
-is applied to all audio files matching that pattern:
-
- para_client setatt test+ '/test/directory/*'
-
-The command
-
- para_client -- ls -lv
-
-gives you a verbose listing of your audio files which contains also
-which attributes are set.
-
-In case you wonder why the double-dash in the above command is needed:
-It tells para_client to not interpret the options after the dashes. If
-you find this annoying, just say
-
- alias para='para_client --'
-
-and be happy. In what follows we shall use this alias.
-
-The "test" attribute can be dropped from the database with
-
- para rmatt test
-
-Read the output of
-
- para help ls
- para help setatt
-
-for more information and a complete list of command line options to
-these commands.
-
-
-----------------------
-Abstract mood nonsense
-~~~~~~~~~~~~~~~~~~~~~~
-
-[skip this part if you don't like formal definitions]
-
-A mood consists of a unique name and its *mood definition*, which is
-a set of *mood lines* containing expressions in terms of attributes
-and other data contained in the database.
-
-A mood defines a subset of audio files called the *admissible audio
-files* for that mood. At any time, at most one mood can be *active*
-which means that para_server is going to select only files from that
-subset of admissible files.
-
-So in order to create a mood definition one has to write a set of
-mood lines. Mood lines come in three flavours: Accept lines, deny
-lines and score lines.
-
-The general syntax of the three types of mood lines is
-
-
- accept [with score <score>] [if] [not] <mood_method> [options]
- deny [with score <score>] [if] [not] <mood_method> [options]
- score <score> [if] [not] <mood_method> [options]
-
-
-Here <score> is either an integer or the string "random" which assigns
-a random score to all matching files. The score value changes the
-order in which admissible files are going to be selected, but is of
-minor importance for this introduction.
-
-So we concentrate on the first two forms, i.e. accept and deny
-lines. As usual, everything in square brackets is optional, i.e.
-accept/deny lines take the following form when ignoring scores:
-
- accept [if] [not] <mood_method> [options]
-
-and analogously for the deny case. The "if" keyword is purely cosmetic
-and has no function. The "not" keyword just inverts the result, so
-the essence of a mood line is the mood method part and the options
-following thereafter.
-
-A *mood method* is realized as a function which takes an audio file
-and computes a number from the data contained in the database.
-If this number is non-negative, we say the file *matches* the mood
-method. The file matches the full mood line if it either
-
- - matches the mood method and the "not" keyword is not given,
-or
- - does not match the mood method, but the "not" keyword is given.
-
-The set of admissible files for the whole mood is now defined as those
-files which match at least one accept mood line, but no deny mood line.
-More formally, an audio file F is admissible if and only if
-
- (F ~ AL1 or F ~ AL2...) and not (F ~ DL1 or F ~ DN2 ...)
-
-where AL1, AL2... are the accept lines, DL1, DL2... are the deny
-lines and "~" means "matches".
-
-The cases where no mood lines of accept/deny type are defined need
-special treatment:
-
- - Neither accept nor deny lines: This treats all files as admissible
- (in fact, that is the definition of the dummy mood which is activated
- automatically if no moods are available).
-
- - Only accept lines: A file is admissible iff it matches at least one
- accept line:
-
- F ~ AL1 or F ~ AL2 or ...
-
- - Only deny lines: A file is admissible iff it matches no deny line:
-
- not (F ~ DL1 or F ~ DN2 ...)
-
-
-
---------------------
-List of mood_methods
-~~~~~~~~~~~~~~~~~~~~
-
- no_attributes_set
-
-Takes no arguments and matches an audio file if and only if no
-attributes are set.
-
- played_rarely
-
-Takes no arguments and matches all audio files where the number of
-times this audio file was selected is below the average.
-
- is_set attribute_name
-
-Takes the name of an attribute and matches iff that attribute is set.
-
- path_matches pattern
-
-Takes a filename pattern and matches iff the path of the audio file
-matches the pattern.
-
-
-----------
-Mood usage
-~~~~~~~~~~
-
-To create a new mood called "my_mood", write its definition into
-some temporary file, say "tmpfile", and add it to the mood table
-by executing
-
- para addmood my_mood < tmpfile
-
-If the mood definition is really short, you may just pipe it to the
-client instead of using temporary files. Like this:
-
- echo "$MOOD_DEFINITION" | para addmood my_mood
-
-There is no need to keep the temporary file since you can always use
-the catmood command to get it back:
-
- para catmood my_mood
-
-A mood can be activated by executing
-
- para select m/my_mood
-
-Once active, the list of admissible files is shown by the ls command
-if the "-a" switch is given:
-
- para ls -a
-
------------------------
-Example mood definition
-~~~~~~~~~~~~~~~~~~~~~~~
-
-Suppose you have defined attributes "punk" and "rock" and want to define
-a mood containing only Punk-Rock songs. That is, an audio file should be
-admissible if and only if both attributes are set. Since
-
- punk and rock
-
-is obviously the same as
-
- not (not punk or not rock)
-
-(de Morgan's rule), a mood definition that selects only Punk-Rock
-songs is
-
- deny if not is_set punk
- deny if not is_set rock
-
-
----------
-Troubles?
----------
-
-Use the debug loglevel (option -l debug for most commands) to show
-debugging info. Almost all paraslash executables have a brief online
-help which is displayed by using the -h switch. The --detailed-help
-option prints the full help text.
-
-para_fsck tries to fix your database. Use --force (even if your name
-isn't Luke) to clean up after a crash. However, first make sure
-para_server isn't running before executing para_fsck if para_fsck
-complains about busy (dirty) tables. para_fsck also contains an option
-to dump the contents of the database to the file system.
-
-If you don't mind to recreate your database you can start
-from scratch by removing the entire database directory, i.e.
-
- rm -rf ~/.paraslash/afs_database
-
-Note that this removes all tables, in particular attribute definitions
-and data, and all playlist and mood definitions.
-
-para_fsck operates on the osl-layer, i.e. it fixes inconsistencies
-in the database but doesn't know about the contents of the tables
-contained therein. Use
-
- para_client check
-
-to print out bad entries, e.g.missing audio files or invalid mood
-definitions.
-
-Still having problems? mailto: Andre Noll <maan@systemlinux.org>
+++ /dev/null
-Requirements
-============
-
-In any case you need
-
- - gcc, the gnu compiler collection (shipped with distro): gcc-3.3
- or newer is required.
- - gnu make (shipped with disto, might be called gmake)
- - bash (most likely already installed)
- - A decent version of grep. Solaris' /bin/grep is not good enough,
- /usr/xpg4/bin/grep is fine though.
- - openssl (needed by server, client): usually shipped with
- distro, but you might have to install the "development package"
- (called libssl-dev on debian systems) as well:
- http://www.openssl.org/
- - help2man (for man page creation) ftp://ftp.gnu.org/pub/gnu/help2man
- - software mixing, e.g. ALSA and the direct mixing plugin (dmix)
-
-Optional features:
-
- - *mp3*: The mp3 decoder of para_filter is based on libmad:
- http://www.underbit.com/products/mad/. If you prefer to
- use the libmad package provided by your distributor, make
- sure to install the corresponding development package as
- well. It is called libmad0-dev on debian-based systems.
- Note that libmad is not necessary for the server side,
- i.e. for sending mp3 files.
-
- - *id3 tags*:
- For version-2 id3 tag support, you'll need libid3tag which
- is also available through the above link (alternatively:
- install package libid3tag0-dev). Without libid3tag, only
- version one tags are recognized.
-
- - *ogg vorbis*: For ogg vorbis streams you'll need libogg,
- libvorbis, libvorbisfile: http://www.xiph.org/downloads/.
- The corresponding Debian packages are called libogg-dev
- libvorbis-dev, other distributors chose similar names.
-
- - *aac*:
- For aac files (m4a) you'll need libfaad. Get it at
- http://www.audiocoding.com/.
- Debian package: libfaad-dev.
-
- - On Linux, you'll need to have ALSA's development package
- installed. The Debian package is called libasound2-dev.
-
-Hacking the source:
-
- - gengetopt: ftp://ftp.gnu.org/pub/gnu/gengetopt/
- - autoconf: ftp://ftp.gnu.org/pub/gnu/autoconf/
- - git http://git.or.cz/
- - grutatxt http://www.triptico.com/software/grutatxt.html
- - doxygen http://www.stack.nl/~dimitri/doxygen/
- - global ftp://ftp.gnu.org/pub/gnu/global
- - m4: ftp://ftp.gnu.org/pub/gnu/m4/
/*
- * Copyright (C) 2006-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2006-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/*
- * Copyright (C) 2006-2008 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2006-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file aac_afh.c para_server's aac audio format handler */
+#include <regex.h>
+
#include "para.h"
#include "error.h"
-#include "string.h"
#include "afh.h"
-#include "afs.h"
-#include "server.h"
+#include "string.h"
#include "aac.h"
static int aac_find_stsz(unsigned char *buf, size_t buflen, off_t *skip)
return buf;
}
-static char *read_tags(unsigned char *buf, size_t buflen)
+static void read_tags(unsigned char *buf, size_t buflen, struct afh_info *afhi)
{
unsigned char *p = buf;
- char *title = NULL, *artist = NULL, *album = NULL, *year = NULL,
- *comment = NULL, *result;
while (p + 32 < buf + buflen) {
unsigned char *q, type1[5], type2[5];
if (q + size2 > buf + buflen)
break;
if (!atom_cmp(type1, "©ART"))
- artist = get_tag(q, size2);
+ afhi->tags.artist = get_tag(q, size2);
else if (!atom_cmp(type1, "©alb"))
- album = get_tag(q, size2);
+ afhi->tags.album = get_tag(q, size2);
else if (!atom_cmp(type1, "©nam"))
- title = get_tag(q, size2);
+ afhi->tags.title = get_tag(q, size2);
else if (!atom_cmp(type1, "©cmt"))
- comment = get_tag(q, size2);
+ afhi->tags.comment = get_tag(q, size2);
else if (!atom_cmp(type1, "©day"))
- year = get_tag(q, size2);
+ afhi->tags.year = get_tag(q, size2);
p += size1;
}
- result = make_taginfo(title, artist, album, year, comment);
- free(title);
- free(artist);
- free(album);
- free(year);
- free(comment);
- return result;
}
-static char *read_meta(unsigned char *buf, size_t buflen)
+static void read_meta(unsigned char *buf, size_t buflen, struct afh_info *afhi)
{
unsigned char *p = buf;
continue;
}
p += 4;
- return read_tags(p, buflen - (p - buf));
+ return read_tags(p, buflen - (p - buf), afhi);
}
- return make_taginfo(NULL, NULL, NULL, NULL, NULL);
}
-static char *aac_get_taginfo(unsigned char *buf, size_t buflen)
+static void aac_get_taginfo(unsigned char *buf, size_t buflen,
+ struct afh_info *afhi)
{
int i;
uint64_t subsize;
p = buf + i;
i += read_atom_header(p, &subsize, type);
p = buf + i;
- return read_meta(p, buflen - i);
+ return read_meta(p, buflen - i, afhi);
}
PARA_INFO_LOG("no meta data\n");
- return make_taginfo(NULL, NULL, NULL, NULL, NULL);
}
static ssize_t aac_compute_chunk_table(struct afh_info *afhi,
mp4AudioSpecificConfig mp4ASC;
NeAACDecHandle handle = NULL;
unsigned char *umap = (unsigned char *) map;
- char *taginfo;
ret = aac_find_esds(umap, numbytes, &skip, &decoder_len);
if (ret < 0)
goto out;
- taginfo = aac_get_taginfo(umap, numbytes);
+ aac_get_taginfo(umap, numbytes, afhi);
handle = aac_open();
ret = -E_AAC_AFH_INIT;
if (NeAACDecInit(handle, umap + skip, decoder_len, &rate, &channels))
ret = (afhi->chunk_table[afhi->chunks_total] - afhi->chunk_table[0]) * 8; /* bits */
ret += (channels * afhi->seconds_total * 500); /* avoid rounding error */
afhi->bitrate = ret / (channels * afhi->seconds_total * 1000);
- afhi->info_string = make_message("%s:\n%s",
- status_item_list[SI_AUDIO_FILE_INFO],
- taginfo);
- free(taginfo);
- tv_scale(20, &afhi->chunk_tv, &afhi->eof_tv);
ret = 1;
out:
if (handle)
/*
- * Copyright (C) 2006-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2006-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/*
- * Copyright (C) 2006-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2006-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file aacdec_filter.c paraslash's aac (m4a) decoder. */
-#include "para.h"
+#include <regex.h>
+#include <stdbool.h>
+#include "para.h"
#include "list.h"
#include "sched.h"
#include "ggo.h"
+#include "buffer_tree.h"
#include "filter.h"
#include "error.h"
#include "string.h"
#include "aac.h"
-/** the output buffer size */
-#define AAC_OUTBUF_SIZE (32 * 1024)
-
-/** give up decoding after that many errors */
+/** Give up decoding after that many errors. */
#define MAX_ERRORS 20
/**
size_t consumed_total;
/** return value of aac_find_entry_point */
size_t entry;
+ /** The number of channels of the current stream. */
+ unsigned int channels;
+ /** Current sample rate in Hz. */
+ unsigned int sample_rate;
};
-static ssize_t aacdec(char *input_buffer, size_t len, struct filter_node *fn)
+static int aacdec_execute(struct btr_node *btrn, const char *cmd, char **result)
+{
+ struct filter_node *fn = btr_context(btrn);
+ struct private_aacdec_data *padd = fn->private_data;
+
+ return decoder_execute(cmd, padd->sample_rate, padd->channels, result);
+}
+
+static void aacdec_open(struct filter_node *fn)
+{
+ struct private_aacdec_data *padd = para_calloc(sizeof(*padd));
+
+ fn->private_data = padd;
+ fn->min_iqs = 2048;
+ padd->handle = aac_open();
+}
+
+static void aacdec_close(struct filter_node *fn)
{
struct private_aacdec_data *padd = fn->private_data;
- struct filter_chain *fc = fn->fc;
+
+ NeAACDecClose(padd->handle);
+ free(padd);
+ fn->private_data = NULL;
+}
+
+static void aacdec_post_select(__a_unused struct sched *s, struct task *t)
+{
+ struct filter_node *fn = container_of(t, struct filter_node, task);
+ struct btr_node *btrn = fn->btrn;
+ struct private_aacdec_data *padd = fn->private_data;
int i, ret;
- unsigned char *p, *outbuffer;
- unsigned char *inbuf = (unsigned char*)input_buffer;
- size_t skip, consumed = 0;
+ unsigned char *p, *inbuf, *outbuffer;
+ char *btr_buf;
+ size_t len, skip, consumed, loaded;
- if (fn->loaded > fn->bufsize * 3 / 5)
- return 0;
- ret = *fc->input_error;
+next_buffer:
+ t->error = 0;
+ ret = btr_node_status(btrn, fn->min_iqs, BTR_NT_INTERNAL);
if (ret < 0)
- return ret;
- if (len < 2048)
- return 0;
-
+ goto err;
+ if (ret == 0)
+ return;
+ btr_merge(btrn, fn->min_iqs);
+ len = btr_next_buffer(btrn, (char **)&inbuf);
+ len = PARA_MIN(len, (size_t)8192);
+ consumed = 0;
if (!padd->initialized) {
unsigned long rate = 0;
unsigned char channels = 0;
&channels) < 0)
goto out;
}
- fc->samplerate = rate;
- fc->channels = channels;
+ padd->sample_rate = rate;
+ padd->channels = channels;
PARA_INFO_LOG("rate: %u, channels: %d\n",
- fc->samplerate, fc->channels);
+ padd->sample_rate, padd->channels);
padd->initialized = 1;
}
if (padd->decoder_length > 0) {
if (consumed >= len)
goto success;
p = inbuf + consumed;
+ //PARA_CRIT_LOG("consumed: %zu (%zu + %zu), have: %zu\n", padd->consumed_total + consumed,
+ // padd->consumed_total, consumed, len - consumed);
outbuffer = NeAACDecDecode(padd->handle, &padd->frame_info, p,
len - consumed);
if (padd->frame_info.error) {
+ int err = padd->frame_info.error;
ret = -E_AAC_DECODE;
if (padd->error_count++ > MAX_ERRORS)
- goto out;
- PARA_ERROR_LOG("frame_error: %d, consumed: %zu + %zd + %lu\n",
- padd->frame_info.error, padd->consumed_total,
- consumed, padd->frame_info.bytesconsumed);
- PARA_ERROR_LOG("%s\n", NeAACDecGetErrorMessage(
- padd->frame_info.error));
- consumed++; /* catch 21 */
+ goto err;
+ /* Suppress non-fatal bitstream error message at BOF/EOF */
+ if (len < fn->min_iqs || padd->consumed_total == 0) {
+ consumed = len;
+ goto success;
+ }
+ PARA_ERROR_LOG("%s\n", NeAACDecGetErrorMessage(err));
+ PARA_ERROR_LOG("consumed: %zu + %zd + %lu\n",
+ padd->consumed_total, consumed,
+ padd->frame_info.bytesconsumed);
+ if (consumed < len)
+ consumed++; /* catch 21 */
goto success;
}
padd->error_count = 0;
+ //PARA_CRIT_LOG("decoder ate %lu\n", padd->frame_info.bytesconsumed);
consumed += padd->frame_info.bytesconsumed;
ret = consumed;
if (!padd->frame_info.samples)
goto out;
- ret = -E_AAC_OVERRUN;
- if (padd->frame_info.samples * 2 + fn->loaded > fn->bufsize)
- goto out;
+ btr_buf = para_malloc(2 * padd->frame_info.samples);
+ loaded = 0;
for (i = 0; i < padd->frame_info.samples; i++) {
- short *s = (short *)outbuffer;
- write_int16_host_endian(fn->buf + fn->loaded, s[i]);
- fn->loaded += 2;
+ short sh = ((short *)outbuffer)[i];
+ write_int16_host_endian(btr_buf + loaded, sh);
+ loaded += 2;
}
+ btr_add_output(btr_buf, loaded, btrn);
success:
ret = consumed;
out:
- if (ret > 0)
+ if (ret >= 0) {
padd->consumed_total += ret;
- return ret;
-}
-
-static void aacdec_open(struct filter_node *fn)
-{
- struct private_aacdec_data *padd = para_calloc(sizeof(*padd));
-
- fn->private_data = padd;
- fn->bufsize = AAC_OUTBUF_SIZE;
- fn->buf = para_calloc(fn->bufsize);
- padd->handle = aac_open();
-}
-
-static void aacdec_close(struct filter_node *fn)
-{
- struct private_aacdec_data *padd = fn->private_data;
-
- NeAACDecClose(padd->handle);
- free(fn->buf);
- fn->buf = NULL;
- free(padd);
- fn->private_data = NULL;
+ btr_consume(btrn, ret);
+ goto next_buffer;
+ }
+err:
+ assert(ret < 0);
+ t->error = ret;
+ btr_remove_node(btrn);
}
/**
- * the init function of the aacdec filter
+ * The init function of the aacdec filter.
*
- * \param f pointer to the filter struct to initialize
+ * \param f Pointer to the filter struct to initialize.
*
* \sa filter::init
*/
void aacdec_filter_init(struct filter *f)
{
f->open = aacdec_open;
- f->convert = aacdec;
f->close = aacdec_close;
+ f->pre_select = generic_filter_pre_select;
+ f->post_select = aacdec_post_select;
+ f->execute = aacdec_execute;
}
/*
- * Copyright (C) 2005-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2005-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file acl.c Access control lists for paraslash senders. */
+#include <regex.h>
+
#include "para.h"
#include "error.h"
#include "string.h"
/*
- * Copyright (C) 2008-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2008-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/*
- * Copyright (C) 2008-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2008-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file afh.c Paraslash's standalone audio format handler tool. */
+#include <regex.h>
#include <dirent.h>
#include <sys/time.h>
"%s: %dHz\n" /* frequency */
"%s: %d\n" /* channels */
"%s: %lu\n" /* seconds total */
- "%s" /* tag info */
"%s: %lu: %lu\n" /* chunk time */
- "%s: %lu\n", /* num chunks */
+ "%s: %lu\n" /* num chunks */
+ "%s: %s\n" /* techinfo */
+ "%s: %s\n" /* artist */
+ "%s: %s\n" /* title */
+ "%s: %s\n" /* year */
+ "%s: %s\n" /* album */
+ "%s: %s\n", /* comment */
status_item_list[SI_BITRATE], afhi->bitrate,
status_item_list[SI_FORMAT], audio_format_name(audio_format_num),
status_item_list[SI_FREQUENCY], afhi->frequency,
status_item_list[SI_CHANNELS], afhi->channels,
status_item_list[SI_SECONDS_TOTAL], afhi->seconds_total,
- afhi->info_string,
status_item_list[SI_CHUNK_TIME], (long unsigned)afhi->chunk_tv.tv_sec,
(long unsigned)afhi->chunk_tv.tv_usec,
- status_item_list[SI_NUM_CHUNKS], afhi->chunks_total
+ status_item_list[SI_NUM_CHUNKS], afhi->chunks_total,
+ status_item_list[SI_TECHINFO], afhi->techinfo? afhi->techinfo : "",
+ status_item_list[SI_ARTIST], afhi->tags.artist? afhi->tags.artist : "",
+ status_item_list[SI_TITLE], afhi->tags.title? afhi->tags.title : "",
+ status_item_list[SI_YEAR], afhi->tags.year? afhi->tags.year : "",
+ status_item_list[SI_ALBUM], afhi->tags.album? afhi->tags.album : "",
+ status_item_list[SI_COMMENT], afhi->tags.comment? afhi->tags.comment : ""
);
}
{
int i;
- printf("chunk_table: ");
- for (i = 0; i <= afhi->chunks_total; i++)
- printf("%u ", afhi->chunk_table[i]);
- printf("\n");
+ if (!conf.human_given) {
+ printf("chunk_table: ");
+ for (i = 0; i <= afhi->chunks_total; i++)
+ printf("%u ", afhi->chunk_table[i]);
+ printf("\n");
+ return;
+ }
+ for (i = 1; i <= afhi->chunks_total; i++) {
+ struct timeval tv;
+ long unsigned from, to;
+ tv_scale(i - 1, &afhi->chunk_tv, &tv);
+ from = tv2ms(&tv);
+ tv_scale(i, &afhi->chunk_tv, &tv);
+ to = tv2ms(&tv);
+ printf("%d [%lu.%03lu - %lu.%03lu] %u - %u (%u)\n", i - 1,
+ from / 1000, from % 1000, to / 1000, to % 1000,
+ afhi->chunk_table[i - 1], afhi->chunk_table[i],
+ afhi->chunk_table[i] - afhi->chunk_table[i - 1]);
+ }
}
static int cat_file(void *audio_file_data, struct afh_info *afhi)
print_info(audio_format_num, &afhi);
if (conf.chunk_table_given)
print_chunk_table(&afhi);
+ free(afhi.techinfo);
+ free(afhi.tags.artist);
+ free(afhi.tags.title);
+ free(afhi.tags.year);
+ free(afhi.tags.album);
+ free(afhi.tags.comment);
+ free(afhi.chunk_table);
printf("\n");
}
- free(afhi.chunk_table);
- free(afhi.info_string);
ret2 = para_munmap(audio_file_data, audio_file_size);
if (ret2 < 0 && ret >= 0)
ret = ret2;
/*
- * Copyright (C) 2005-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2005-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file afh.h structures for audio format handling (para_server) */
-/** \cond */
-#ifdef HAVE_OGGVORBIS
-#define OV_AUDIO_FORMAT " ogg"
-#else
-#define OV_AUDIO_FORMAT ""
-#endif
-
-#ifdef HAVE_FAAD
-#define AAC_AUDIO_FORMAT " aac"
-#else
-#define AAC_AUDIO_FORMAT ""
-#endif
-
-#define SUPPORTED_AUDIO_FORMATS "mp3" OV_AUDIO_FORMAT AAC_AUDIO_FORMAT
-
-/** \endcond */
+/**
+ * The tags used by all audio format handlers.
+ *
+ * Paraslash only uses the more common tags. These are recognized
+ * for all supported audio formats.
+ */
+struct taginfo {
+ /** TPE1 (id3v2) / ARTIST (vorbis) / ©ART (aac) */
+ char *artist;
+ /** TIT2/TITLE/©nam */
+ char *title;
+ /** TDRC/YEAR/©day */
+ char *year;
+ /** TALB/ALBUM/©alb */
+ char *album;
+ /** COMM/COMMENT/©cmt */
+ char *comment;
+};
/** Audio format dependent information. */
struct afh_info {
long unsigned chunks_total;
/** The length of the audio file in seconds. */
long unsigned seconds_total;
- /** A string that gets filled in by the audio format handler. */
- char *info_string;
+ /** Audio handler specific info about the file. */
+ char *techinfo;
+ /** Id3 tags, vorbis comments, aac tags. */
+ struct taginfo tags;
/**
* The table that specifies the offset of the individual pieces in
* the current audio file.
uint32_t *chunk_table;
/** Period of time between sending data chunks. */
struct timeval chunk_tv;
- /** End of file timeout - Do not load new audio file until this time. */
- struct timeval eof_tv;
/**
* The position of the header within the audio file. Ignored if \a
* header_len equals zero.
const char *audio_format_name(int);
void afh_get_chunk(long unsigned chunk_num, struct afh_info *afhi,
void *map, const char **buf, size_t *len);
-uint32_t afh_get_largest_chunk_size(struct afh_info *afhi);
void afh_get_header(struct afh_info *afhi, void *map, const char **buf, size_t *len);
-char *make_taginfo(char *title, char *artist, char *album, char *year,
- char *comment);
/*
- * Copyright (C) 1997-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 1997-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
#include <sys/time.h> /* gettimeofday */
#include <sys/types.h>
#include <dirent.h>
+#include <regex.h>
#include "para.h"
#include "error.h"
#ifdef HAVE_FAAD
void aac_afh_init(struct audio_format_handler *);
#endif
+#ifdef HAVE_SPEEX
+ void spx_afh_init(struct audio_format_handler *);
+#endif
+void wma_afh_init(struct audio_format_handler *);
/**
* The list of supported audio formats.
*
.name = "aac",
#ifdef HAVE_FAAD
.init = aac_afh_init,
+#endif
+ },
+ {
+ .name = "wma",
+ .init = wma_afh_init,
+ },
+ {
+ .name = "spx",
+#ifdef HAVE_SPEEX
+ .init = spx_afh_init,
#endif
},
{
int i;
PARA_INFO_LOG("supported audio formats: %s\n",
- SUPPORTED_AUDIO_FORMATS);
+ SERVER_AUDIO_FORMATS);
FOR_EACH_AUDIO_FORMAT(i) {
PARA_NOTICE_LOG("initializing %s handler\n",
audio_format_name(i));
return -E_AUDIO_FORMAT;
}
-/**
- * Pretty-print the given meta-info.
- *
- * \param title The title of the audio file.
- * \param artist The artist.
- * \param album The name of the album.
- * \param year Year of release.
- * \param comment Further comments.
- *
- * This function is called by each audio format handler to produce the tag info
- * status items. Usually, the audio format handlers read this info from the
- * audio file (id3 tags, vorbis comments, ...).
- *
- * It is OK to pass \p NULL pointers for any argument in which case a suitable
- * string is inserted which indicates that this information is not available.
- *
- * \return The status item string. It must be freed by the caller.
- */
-char *make_taginfo(char *title, char *artist, char *album, char *year,
- char *comment)
-{
- return make_message("%s: %s, by %s\n" /* taginfo1 */
- "%s: A: %s, Y: %s, C: %s\n", /* taginfo2 */
- status_item_list[SI_TAGINFO1],
- (title && *title)? title : "(title tag not set)",
- (artist && *artist)? artist : "(artist tag not set)",
- status_item_list[SI_TAGINFO2],
- (album && *album)? album : "(album tag not set)",
- (year && *year)? year : "????",
- (comment && *comment)? comment : "(comment tag not set)"
- );
-}
-
/**
* Call get_file_info() to obtain an afhi structure.
*
afhi->header_offset = 0;
afhi->header_len = 0;
+ afhi->techinfo = NULL;
+ afhi->tags.artist = NULL;
+ afhi->tags.title = NULL;
+ afhi->tags.year = NULL;
+ afhi->tags.album = NULL;
+ afhi->tags.comment = NULL;
format = guess_audio_format(path);
if (format >= 0) {
ret = afl[format].get_file_info(data, size, fd, afhi);
- if (ret >= 0)
- return format;
+ if (ret >= 0) {
+ ret = format;
+ goto success;
+ }
}
FOR_EACH_AUDIO_FORMAT(i) {
if (i == format) /* we already tried this one to no avail */
continue;
ret = afl[i].get_file_info(data, size, fd, afhi);
- if (ret >= 0)
- return i;
+ if (ret >= 0) {
+ ret = i;
+ goto success;
+ }
PARA_WARNING_LOG("%s\n", para_strerror(-ret));
}
return -E_AUDIO_FORMAT;
+success:
+ if (!afhi->techinfo)
+ afhi->techinfo = para_strdup(NULL);
+ if (!afhi->tags.artist)
+ afhi->tags.artist = para_strdup(NULL);
+ if (!afhi->tags.title)
+ afhi->tags.title = para_strdup(NULL);
+ if (!afhi->tags.year)
+ afhi->tags.year = para_strdup(NULL);
+ if (!afhi->tags.album)
+ afhi->tags.album = para_strdup(NULL);
+ if (!afhi->tags.comment)
+ afhi->tags.comment = para_strdup(NULL);
+ PARA_DEBUG_LOG("techinfo: %s\n", afhi->techinfo);
+ PARA_DEBUG_LOG("artist: %s\n", afhi->tags.artist);
+ PARA_DEBUG_LOG("title: %s\n", afhi->tags.title);
+ PARA_DEBUG_LOG("year: %s\n", afhi->tags.year);
+ PARA_DEBUG_LOG("album: %s\n", afhi->tags.album);
+ PARA_DEBUG_LOG("comment: %s\n", afhi->tags.comment);
+ return ret;
}
/**
*len = afhi->chunk_table[chunk_num + 1] - pos;
}
-uint32_t afh_get_largest_chunk_size(struct afh_info *afhi)
-{
- uint32_t n, largest = 0, *ct = afhi->chunk_table;
-
- for (n = 1; n <= afhi->chunks_total; n++)
- largest = PARA_MAX(largest, ct[n] - ct[n - 1]);
- return largest;
-}
-
/**
* Get the header of an audio file.
*
*
* This function sets \a buf to \p NULL and \a len to zero if \a map or \a
* afhi is \p NULL, or if the current audio format does not need special
- * header treamtment.
+ * header treatment.
*/
void afh_get_header(struct afh_info *afhi, void *map, const char **buf, size_t *len)
{
/*
- * Copyright (C) 2007-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2007-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file afs.c Paraslash's audio file selector. */
+#include <regex.h>
#include <signal.h>
#include <fnmatch.h>
+#include <openssl/rc4.h>
+#include <osl.h>
+
#include "server.cmdline.h"
#include "para.h"
#include "error.h"
+#include "crypt.h"
#include "string.h"
#include "afh.h"
#include "afs.h"
#include "sched.h"
#include "signal.h"
#include "fd.h"
+#include "mood.h"
/** The osl tables used by afs. \sa blob.c. */
enum afs_table_num {
static enum play_mode current_play_mode;
static char *current_mop; /* mode or playlist specifier. NULL means dummy mooe */
-
/**
* A random number used to "authenticate" the connection.
*
*(uint32_t *) buf = afs_socket_cookie;
*(int *) (buf + sizeof(afs_socket_cookie)) = query_shmid;
- ret = create_remote_socket(conf.afs_socket_arg);
+ ret = connect_local_socket(conf.afs_socket_arg);
if (ret < 0)
goto out;
fd = ret;
struct pattern_match_data *pmd = data;
struct osl_object name_obj;
const char *p, *name;
- int ret = osl_get_object(pmd->table, row, pmd->match_col_num, &name_obj);
+ int ret = osl(osl_get_object(pmd->table, row, pmd->match_col_num, &name_obj));
const char *pattern_txt = (const char *)pmd->patterns.data;
if (ret < 0)
*
* \param pmd Describes what to match and how.
*
- * \return The return value of the underlying call to osl_rbtree_loop()
- * or osl_rbtree_loop_reverse().
+ * \return Standard.
*/
int for_each_matching_row(struct pattern_match_data *pmd)
{
if (pmd->pm_flags & PM_REVERSE_LOOP)
- return osl_rbtree_loop_reverse(pmd->table, pmd->loop_col_num, pmd,
- action_if_pattern_matches);
- return osl_rbtree_loop(pmd->table, pmd->loop_col_num, pmd,
- action_if_pattern_matches);
+ return osl(osl_rbtree_loop_reverse(pmd->table, pmd->loop_col_num, pmd,
+ action_if_pattern_matches));
+ return osl(osl_rbtree_loop(pmd->table, pmd->loop_col_num, pmd,
+ action_if_pattern_matches));
}
/**
return strncmp(str1, str2, PARA_MIN(obj1->size, obj2->size));
}
-/*
- * write input from fd to dynamically allocated buffer,
- * but maximal max_size byte.
- */
-static int fd2buf(int fd, unsigned max_size, struct osl_object *obj)
-{
- const size_t chunk_size = 1024;
- size_t size = 2048, received = 0;
- int ret;
- char *buf = para_malloc(size);
-
- for (;;) {
- ret = recv_bin_buffer(fd, buf + received, chunk_size);
- if (ret <= 0)
- break;
- received += ret;
- if (received + chunk_size >= size) {
- size *= 2;
- ret = -E_INPUT_TOO_LARGE;
- if (size > max_size)
- break;
- buf = para_realloc(buf, size);
- }
- }
- obj->data = buf;
- obj->size = received;
- if (ret < 0)
- free(buf);
- return ret;
-}
-
-/**
- * Read data from a file descriptor, and send it to the afs process.
- *
- * \param fd File descriptor to read data from.
- * \param arg_obj Pointer to the arguments to \a f.
- * \param f The callback function.
- * \param max_len Don't read more than that many bytes from stdin.
- * \param result_handler See \ref send_callback_request.
- * \param private_result_data See \ref send_callback_request.
- *
- * This function is used by commands that wish to let para_server store
- * arbitrary data specified by the user (for instance the add_blob family of
- * commands). First, at most \a max_len bytes are read from \a fd, the result
- * is concatenated with the buffer given by \a arg_obj, and the combined buffer
- * is made available to the afs process via the callback method. See \ref
- * send_callback_request for details.
- *
- * \return Negative on errors, the return value of the underlying call to
- * send_callback_request() otherwise.
- */
-int stdin_command(int fd, struct osl_object *arg_obj, callback_function *f,
- unsigned max_len, callback_result_handler *result_handler,
- void *private_result_data)
-{
- struct osl_object query, stdin_obj;
- int ret;
-
- ret = send_buffer(fd, AWAITING_DATA_MSG);
- if (ret < 0)
- return ret;
- ret = fd2buf(fd, max_len, &stdin_obj);
- if (ret < 0)
- return ret;
- query.size = arg_obj->size + stdin_obj.size;
- query.data = para_malloc(query.size);
- memcpy(query.data, arg_obj->data, arg_obj->size);
- memcpy((char *)query.data + arg_obj->size, stdin_obj.data, stdin_obj.size);
- free(stdin_obj.data);
- ret = send_callback_request(f, &query, result_handler, private_result_data);
- free(query.data);
- return ret;
-}
-
static int pass_afd(int fd, char *buf, size_t size)
{
struct msghdr msg = {.msg_iov = NULL};
* Result handler for sending data to the para_client process.
*
* \param result The data to be sent.
- * \param fd_ptr Pointer to the file descriptor.
+ * \param private Pointer to rc4 context.
*
- * \return The return value of the underlying call to send_bin_buffer().
+ * \return The return value of the underlying call to rc4_send_bin_buffer().
*
- * \sa \ref callback_result_handler.
+ * \sa \ref callback_result_handler, \ref rc4_send_bin_buffer().
*/
-int send_result(struct osl_object *result, void *fd_ptr)
+int rc4_send_result(struct osl_object *result, void *private)
{
- int fd = *(int *)fd_ptr;
+ struct rc4_context *rc4c = private;
+
if (!result->size)
return 1;
- return send_bin_buffer(fd, result->data, result->size);
+ return rc4_send_bin_buffer(rc4c, result->data, result->size);
}
-int com_select(int fd, int argc, char * const * const argv)
+int com_select(struct rc4_context *rc4c, int argc, char * const * const argv)
{
struct osl_object query;
query.data = argv[1];
query.size = strlen(argv[1]) + 1;
return send_callback_request(com_select_callback, &query,
- &send_result, &fd);
+ &rc4_send_result, rc4c);
}
static void init_admissible_files(char *arg)
else {
char *home = para_homedir();
database_dir = make_message(
- "%s/.paraslash/afs_database", home);
+ "%s/.paraslash/afs_database-0.4", home);
free(home);
}
}
static void afs_signal_post_select(struct sched *s, struct task *t)
{
- struct signal_task *st = container_of(t, struct signal_task, task);
+ int signum;
+
if (getppid() == 1) {
PARA_EMERG_LOG("para_server died\n");
goto shutdown;
}
- if (!FD_ISSET(st->fd, &s->rfds))
+ signum = para_next_signal(&s->rfds);
+ if (signum == 0)
return;
- st->signum = para_next_signal();
- if (st->signum == SIGHUP) {
+ if (signum == SIGHUP) {
close_afs_tables();
parse_config_or_die(1);
t->error = open_afs_tables();
init_admissible_files(current_mop);
return;
}
- PARA_EMERG_LOG("terminating on signal %d\n", st->signum);
+ PARA_EMERG_LOG("terminating on signal %d\n", signum);
shutdown:
sched_shutdown();
t->error = -E_AFS_SIGNAL;
return shm_detach(query_shm);
}
-static int execute_server_command(void)
+static int execute_server_command(fd_set *rfds)
{
char buf[8];
- int ret = recv_bin_buffer(server_socket, buf, sizeof(buf) - 1);
+ size_t n;
+ int ret = read_nonblock(server_socket, buf, sizeof(buf) - 1, rfds, &n);
- if (ret <= 0) {
- if (!ret)
- ret = -ERRNO_TO_PARA_ERROR(ECONNRESET);
- goto err;
- }
- buf[ret] = '\0';
- PARA_DEBUG_LOG("received: %s\n", buf);
- ret = -E_BAD_CMD;
+ if (ret < 0 || n == 0)
+ return ret;
+ buf[n] = '\0';
if (strcmp(buf, "new"))
- goto err;
- ret = open_next_audio_file();
-err:
- return ret;
+ return -E_BAD_CMD;
+ return open_next_audio_file();
}
-static void execute_afs_command(int fd, uint32_t expected_cookie)
+/* returns 0 if no data available, 1 else */
+static int execute_afs_command(int fd, fd_set *rfds, uint32_t expected_cookie)
{
uint32_t cookie;
int query_shmid;
char buf[sizeof(cookie) + sizeof(query_shmid)];
- int ret = recv_bin_buffer(fd, buf, sizeof(buf));
+ size_t n;
+ int ret = read_nonblock(fd, buf, sizeof(buf), rfds, &n);
if (ret < 0)
goto err;
- if (ret != sizeof(buf)) {
+ if (n == 0)
+ return 0;
+ if (n != sizeof(buf)) {
PARA_NOTICE_LOG("short read (%d bytes, expected %lu)\n",
ret, (long unsigned) sizeof(buf));
- return;
+ return 1;
}
cookie = *(uint32_t *)buf;
if (cookie != expected_cookie) {
- PARA_NOTICE_LOG("received invalid cookie(got %u, expected %u)\n",
+ PARA_NOTICE_LOG("received invalid cookie (got %u, expected %u)\n",
(unsigned)cookie, (unsigned)expected_cookie);
- return;
+ return 1;
}
query_shmid = *(int *)(buf + sizeof(cookie));
if (query_shmid < 0) {
PARA_WARNING_LOG("received invalid query shmid %d)\n",
query_shmid);
- return;
+ return 1;
}
ret = call_callback(fd, query_shmid);
if (ret >= 0)
- return;
+ return 1;
err:
PARA_NOTICE_LOG("%s\n", para_strerror(-ret));
+ return 1;
}
/** Shutdown connection if query has not arrived until this many seconds. */
struct afs_client *client, *tmp;
int fd, ret;
- if (FD_ISSET(server_socket, &s->rfds)) {
- ret = execute_server_command();
- if (ret < 0) {
- PARA_EMERG_LOG("%s\n", para_strerror(-ret));
- sched_shutdown();
- return;
- }
+ ret = execute_server_command(&s->rfds);
+ if (ret < 0) {
+ PARA_EMERG_LOG("%s\n", para_strerror(-ret));
+ sched_shutdown();
+ return;
}
-
/* Check the list of connected clients. */
list_for_each_entry_safe(client, tmp, &afs_client_list, node) {
- if (FD_ISSET(client->fd, &s->rfds))
- execute_afs_command(client->fd, ct->cookie);
- else { /* prevent bogus connection flooding */
+ ret = execute_afs_command(client->fd, &s->rfds, ct->cookie);
+ if (ret == 0) { /* prevent bogus connection flooding */
struct timeval diff;
tv_diff(now, &client->connect_time, &diff);
if (diff.tv_sec < AFS_CLIENT_TIMEOUT)
free(client);
}
/* Accept connections on the local socket. */
- if (!FD_ISSET(ct->fd, &s->rfds))
- return;
- ret = para_accept(ct->fd, &unix_addr, sizeof(unix_addr));
- if (ret < 0) {
+ ret = para_accept(ct->fd, &s->rfds, &unix_addr, sizeof(unix_addr), &fd);
+ if (ret < 0)
PARA_NOTICE_LOG("%s\n", para_strerror(-ret));
+ if (ret <= 0)
return;
- }
- fd = ret;
ret = mark_fd_nonblocking(fd);
if (ret < 0) {
PARA_NOTICE_LOG("%s\n", para_strerror(-ret));
free(pb.buf);
}
-int com_init(int fd, int argc, char * const * const argv)
+int com_init(struct rc4_context *rc4c, int argc, char * const * const argv)
{
int i, j, ret;
uint32_t table_mask = (1 << (NUM_AFS_TABLES + 1)) - 1;
return -E_BAD_TABLE_NAME;
}
}
- ret = send_callback_request(create_tables_callback, &query, &send_result, &fd);
+ ret = send_callback_request(create_tables_callback, &query,
+ rc4_send_result, rc4c);
if (ret < 0)
- return send_va_buffer(fd, "%s\n", para_strerror(-ret));
+ return rc4_send_va_buffer(rc4c, "%s\n", para_strerror(-ret));
return ret;
}
CHECK_PLAYLISTS = 4
};
-int com_check(int fd, int argc, char * const * const argv)
+int com_check(struct rc4_context *rc4c, int argc, char * const * const argv)
{
unsigned flags = 0;
int i, ret;
if (!flags)
flags = ~0U;
if (flags & CHECK_AFT) {
- ret = send_callback_request(aft_check_callback, NULL, send_result, &fd);
+ ret = send_callback_request(aft_check_callback, NULL,
+ rc4_send_result, rc4c);
if (ret < 0)
return ret;
}
if (flags & CHECK_PLAYLISTS) {
- ret = send_callback_request(playlist_check_callback, NULL, send_result, &fd);
+ ret = send_callback_request(playlist_check_callback,
+ NULL, rc4_send_result, rc4c);
if (ret < 0)
return ret;
}
if (flags & CHECK_MOODS) {
- ret = send_callback_request(mood_check_callback, NULL, send_result, &fd);
+ ret = send_callback_request(mood_check_callback, NULL,
+ rc4_send_result, rc4c);
if (ret < 0)
return ret;
}
}
}
+/**
+ * Dummy event handler for the images table.
+ *
+ * \param event Unused.
+ * \param pb Unused.
+ * \param data Unused.
+ *
+ * This table does not honor events.
+ */
int images_event_handler(__a_unused enum afs_events event,
__a_unused struct para_buffer *pb, __a_unused void *data)
{
return 1;
}
+/**
+ * Dummy event handler for the lyrics table.
+ *
+ * \param event Unused.
+ * \param pb Unused.
+ * \param data Unused.
+ *
+ * This table does not honor events.
+ */
int lyrics_event_handler(__a_unused enum afs_events event,
__a_unused struct para_buffer *pb, __a_unused void *data)
{
HC: Prototypes for the commands of the audio file selector.
CC: Array of commands for the audio file selector.
AT: server_command
-IN: para error string afh afs server list user_list
+SI: openssl/rc4 osl regex
+IN: para error crypt command string afh afs server list user_list
SN: list of afs commands
TM: mood lyr img pl
---
H:
H: -lv: verbose listing mode
H:
+H: -lp: parser-friendly mode
+H:
H: -lm: mbox listing mode
H:
H: -lc: chunk-table listing mode
---
N: lsatt
P: AFS_READ
-D: List attributes
+D: List attributes.
U: lsatt [-i] [-l] [-r] [pattern]
H: Print the list of all defined attributes which match the
H: given pattern. If no pattern is given, the full list is
N: touch
P: AFS_READ | AFS_WRITE
D: Manipulate the afs data for all audio files matching a pattern.
-U: touch [-n numplayed] [-l lastplayed] [-y lyrics_id] [-i image_id] [-a amp] [-v] [-p] pattern
+U: touch [-n=numplayed] [-l=lastplayed] [-y=lyrics_id] [-i=image_id] [-a=amp] [-v] [-p] pattern
H: If no option is given, lastplayed is set to the current time
H: and numplayed is increased by one. Otherwise, only the given
H: options are taken into account.
H: -l Set lastplayed time. The last time this audio file was selected.
H: Must be given as the number of seconds since the epoch. Example:
H:
-H: touch -l $(date +%s) file
+H: touch -l=$(date +%s) file
H:
H: sets the lastplayed time of 'file' to the current time.
H:
---
T: add
N: add@member@
-O: int com_add@member@(int fd, int argc, char * const * const argv);
+O: int com_add@member@(struct rc4_context *rc4c, int argc, char * const * const argv);
P: AFS_READ | AFS_WRITE
D: Read data from stdin and add it as a blob to the @member@ table.
U: add@member@ @member@_name
---
T: cat
N: cat@member@
-O: int com_cat@member@(int fd, int argc, char * const * const argv);
+O: int com_cat@member@(struct rc4_context *rc4c, int argc, char * const * const argv);
P: AFS_READ
D: Dump the contents of a blob of type @member@ to stdout.
U: cat@member@ @member@_name
---
T: ls
N: ls@member@
-O: int com_ls@member@(int fd, int argc, char * const * const argv);
+O: int com_ls@member@(struct rc4_context *rc4c, int argc, char * const * const argv);
P: AFS_READ
D: List blobs of type @member@ matching a pattern.
U: ls@member@ [-i] [-l] [-r] [pattern]
---
T: rm
N: rm@member@
-O: int com_rm@member@(int fd, int argc, char * const * const argv);
+O: int com_rm@member@(struct rc4_context *rc4c, int argc, char * const * const argv);
P: AFS_READ | AFS_WRITE
D: Remove blob(s) of type @member@ from the @member@ table.
U: rm@member@ pattern...
---
T: mv
N: mv@member@
-O: int com_mv@member@(int fd, int argc, char * const * const argv);
+O: int com_mv@member@(struct rc4_context *rc4c, int argc, char * const * const argv);
P: AFS_READ | AFS_WRITE
D: Rename a blob of type @member@.
U: mv@member@ old_@member@_name new_@member@_name
/*
- * Copyright (C) 2007-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2007-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file afs.h Exported symbols of the audio file selector. */
#include <regex.h>
-#include "osl.h"
#include "hash.h"
/** Audio file selector data stored in the audio file table. */
void *data);
};
-enum play_mode {PLAY_MODE_MOOD, PLAY_MODE_PLAYLIST};
+/** How audio files are selected by afs. */
+enum play_mode {
+ /** Admissible files are determined by a mood definition. */
+ PLAY_MODE_MOOD,
+ /** All listed files are admissible. */
+ PLAY_MODE_PLAYLIST,
+};
/**
* Data about one audio file.
HASH_TYPE *hash;
};
-void make_empty_status_items(char *buf);
-
-/** At most that many bytes will be passed from afs to para_server. */
-#define VERBOSE_LS_OUTPUT_SIZE 4096
-
/** Data about the current audio file, passed from afs to server. */
struct audio_file_data {
- /** Same info as ls -lv -p current audio_file. */
- char verbose_ls_output[VERBOSE_LS_OUTPUT_SIZE];
/** The open file descriptor to the current audio file. */
int fd;
/** Vss needs this for streaming. */
struct afh_info afhi;
+ /** Size of the largest chunk. */
+ uint32_t max_chunk_size;
};
+/**
+ * Codes used for communication between the server and the afs process.
+ *
+ * Before forking the afs child, para_server creates a bidirectional pipe
+ * through which both processes communicate. Usually para_server requests a new
+ * audio in order to start streaming or when the end of the current audio file
+ * has been reached. The afs process responds to such a request by sending
+ * back an eight byte buffer. The first four bytes is the uint32_t
+ * representation of the code, usually \p NEXT_AUDIO_FILE if an admissible
+ * audio file was found, successfully opened and verified. The other four bytes
+ * represent the shared memory id of the shared memory area that contains
+ * details about the audio file to be streamed next. The open file descriptor
+ * of that file is also passed from afs to para_server through the same pipe.
+ */
enum afs_server_code {
+ /** An audio file was successfully opened. */
NEXT_AUDIO_FILE,
+ /** No admissible audio file was found. */
NO_ADMISSIBLE_FILES,
- AFD_CHANGE
};
/** Flags passed to for_each_matching_row(). */
* \sa \ref send_callback_request().
*/
typedef int callback_result_handler(struct osl_object *result, void *private);
-int send_result(struct osl_object *result, void *fd_ptr);
+int rc4_send_result(struct osl_object *result, void *private);
int pass_buffer_as_shm(char *buf, size_t size, void *fd_ptr);
__noreturn void afs_init(uint32_t cookie, int socket_fd);
int send_standard_callback_request(int argc, char * const * const argv,
callback_function *f, callback_result_handler *result_handler,
void *private_result_data);
-int stdin_command(int fd, struct osl_object *arg_obj, callback_function *f,
- unsigned max_len, callback_result_handler *result_handler,
- void *private_result_data);
int string_compare(const struct osl_object *obj1, const struct osl_object *obj2);
int for_each_matching_row(struct pattern_match_data *pmd);
int audio_file_loop(void *private_data, osl_rbtree_loop_func *func);
void aft_check_callback(int fd, __a_unused const struct osl_object *query);
-/* mood */
-int change_current_mood(char *mood_name);
-void close_current_mood(void);
-int reload_current_mood(void);
-void mood_check_callback(int fd, __a_unused const struct osl_object *query);
-
-
/* playlist */
int playlist_open(char *name);
void playlist_close(void);
/** A blob table has that many columns. */
NUM_BLOB_COLUMNS
};
-
-/** Define an osl table description for a blob table. */
-#define DEFINE_BLOB_TABLE_DESC(table_name) \
- struct osl_table_description table_name ## _table_desc = { \
- .name = #table_name, \
- .num_columns = NUM_BLOB_COLUMNS, \
- .flags = OSL_LARGE_TABLE, \
- .column_descriptions = blob_cols \
- };
-
-/** Define a pointer to an osl blob table with a canonical name. */
-#define DEFINE_BLOB_TABLE_PTR(table_name) struct osl_table *table_name ## _table;
-
-/** Define a blob table. */
-#define INIT_BLOB_TABLE(table_name) \
- DEFINE_BLOB_TABLE_DESC(table_name); \
- DEFINE_BLOB_TABLE_PTR(table_name);
-
/*
- * Copyright (C) 2007-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2007-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file aft.c Audio file table functions. */
+#include <regex.h>
#include <dirent.h> /* readdir() */
-#include "para.h"
-#include "error.h"
-#include "string.h"
+#include <openssl/rc4.h>
#include <sys/mman.h>
#include <fnmatch.h>
#include <sys/shm.h>
+#include <osl.h>
+#include "para.h"
+#include "error.h"
+#include "crypt.h"
+#include "string.h"
#include "afh.h"
#include "afs.h"
#include "net.h"
-#include "vss.h"
#include "fd.h"
#include "ipc.h"
#include "portable_io.h"
static struct osl_table *audio_file_table;
+static char *status_items;
+static char *parser_friendly_status_items;
/** The different sorting methods of the ls command. */
enum ls_sorting_method {
/** -lm */
LS_MODE_MBOX,
/** -lc */
- LS_MODE_CHUNKS
+ LS_MODE_CHUNKS,
+ /** -lp */
+ LS_MODE_PARSER,
};
/** The flags accepted by the ls command. */
NUM_AFT_COLUMNS
};
+/**
+ * Compare two osl objects pointing to hash values.
+ *
+ * \param obj1 Pointer to the first hash object.
+ * \param obj2 Pointer to the second hash object.
+ *
+ * \return The values required for an osl compare function.
+ *
+ * \sa osl_compare_func, uint32_compare().
+ */
+static int aft_hash_compare(const struct osl_object *obj1, const struct osl_object *obj2)
+{
+ return hash_compare((HASH_TYPE *)obj1->data, (HASH_TYPE *)obj2->data);
+}
+
static struct osl_column_description aft_cols[] = {
[AFTCOL_HASH] = {
.storage_type = OSL_MAPPED_STORAGE,
.storage_flags = OSL_RBTREE | OSL_FIXED_SIZE | OSL_UNIQUE,
.name = "hash",
- .compare_function = osl_hash_compare,
+ .compare_function = aft_hash_compare,
.data_size = HASH_SIZE
},
[AFTCOL_PATH] = {
CHUNK_TV_TV_USEC_OFFSET = 36,
/** Number of channels is stored here. (1 byte) */
AFHI_CHANNELS_OFFSET = 40,
- /** EOF timeout in ms. (2 byte) */
- AFHI_EOF_OFFSET = 41,
/** The tag info position. */
- AFHI_INFO_STRING_OFFSET = 43,
+ AFHI_INFO_STRING_OFFSET = 41,
/** Minimal on-disk size of a valid afhi struct. */
- MIN_AFHI_SIZE = 44
+ MIN_AFHI_SIZE = 47, /* at least 6 null bytes for techinfo/tags */
};
static unsigned sizeof_afhi_buf(const struct afh_info *afhi)
{
if (!afhi)
return 0;
- return strlen(afhi->info_string) + MIN_AFHI_SIZE;
+ return MIN_AFHI_SIZE
+ + strlen(afhi->techinfo)
+ + strlen(afhi->tags.artist)
+ + strlen(afhi->tags.title)
+ + strlen(afhi->tags.year)
+ + strlen(afhi->tags.album)
+ + strlen(afhi->tags.comment);
}
static void save_afhi(struct afh_info *afhi, char *buf)
{
+ char *p;
+
if (!afhi)
return;
write_u32(buf + AFHI_SECONDS_TOTAL_OFFSET, afhi->seconds_total);
write_u32(buf + HEADER_OFFSET_OFFSET, afhi->header_offset);
write_u32(buf + CHUNK_TV_TV_SEC_OFFSET, afhi->chunk_tv.tv_sec);
write_u32(buf + CHUNK_TV_TV_USEC_OFFSET, afhi->chunk_tv.tv_usec);
- write_u16(buf + AFHI_EOF_OFFSET, tv2ms(&afhi->eof_tv));
- strcpy(buf + AFHI_INFO_STRING_OFFSET, afhi->info_string); /* OK */
+ p = buf + AFHI_INFO_STRING_OFFSET;
+ /* The sprintf's below are OK as our caller made sure that buf is large enough */
+ p += sprintf(p, "%s", afhi->techinfo) + 1;
+ p += sprintf(p, "%s", afhi->tags.artist) + 1;
+ p += sprintf(p, "%s", afhi->tags.title) + 1;
+ p += sprintf(p, "%s", afhi->tags.year) + 1;
+ p += sprintf(p, "%s", afhi->tags.album) + 1;
+ sprintf(p, "%s", afhi->tags.comment);
}
static void load_afhi(const char *buf, struct afh_info *afhi)
afhi->header_offset = read_u32(buf + HEADER_OFFSET_OFFSET);
afhi->chunk_tv.tv_sec = read_u32(buf + CHUNK_TV_TV_SEC_OFFSET);
afhi->chunk_tv.tv_usec = read_u32(buf + CHUNK_TV_TV_USEC_OFFSET);
- ms2tv(read_u16(buf + AFHI_EOF_OFFSET), &afhi->eof_tv);
- afhi->info_string = para_strdup(buf + AFHI_INFO_STRING_OFFSET);
+ afhi->techinfo = (char *)buf + AFHI_INFO_STRING_OFFSET;
+ afhi->tags.artist = afhi->techinfo + strlen(afhi->techinfo) + 1;
+ afhi->tags.title = afhi->tags.artist + strlen(afhi->tags.artist) + 1;
+ afhi->tags.year = afhi->tags.title + strlen(afhi->tags.title) + 1;
+ afhi->tags.album = afhi->tags.year + strlen(afhi->tags.year) + 1;
+ afhi->tags.comment = afhi->tags.album + strlen(afhi->tags.album) + 1;
}
static unsigned sizeof_chunk_table(struct afh_info *afhi)
return 4 * (afhi->chunks_total + 1);
}
-static void save_chunk_table(struct afh_info *afhi, char *buf)
+static uint32_t save_chunk_table(struct afh_info *afhi, char *buf)
{
int i;
-
- for (i = 0; i <= afhi->chunks_total; i++)
- write_u32(buf + 4 * i, afhi->chunk_table[i]);
+ uint32_t max = 0, old = 0;
+
+ for (i = 0; i <= afhi->chunks_total; i++) {
+ uint32_t val = afhi->chunk_table[i];
+ write_u32(buf + 4 * i, val);
+ /*
+ * If the first chunk is the header, do not consider it for the
+ * calculation of the largest chunk size.
+ */
+ if (i == 0 || (i == 1 && afhi->header_len > 0)) {
+ old = val;
+ continue;
+ }
+ max = PARA_MAX(max, val - old);
+ old = val;
+ }
+ return max;
}
static void load_chunk_table(struct afh_info *afhi, char *buf)
* \param path The full path of the audio file.
* \param row Result pointer.
*
- * \return The return value of the underlying call to osl_get_row().
+ * \return Standard.
*/
int aft_get_row_of_path(const char *path, struct osl_row **row)
{
struct osl_object obj = {.data = (char *)path, .size = strlen(path) + 1};
- return osl_get_row(audio_file_table, AFTCOL_PATH, &obj, row);
+ return osl(osl_get_row(audio_file_table, AFTCOL_PATH, &obj, row));
}
/**
* Get the row of the audio file table corresponding to the given hash value.
*
* \param hash The hash value of the desired audio file.
- * \param row resul pointer.
+ * \param row Result pointer.
*
- * \return The return value of the underlying call to osl_get_row().
+ * \return Standard.
*/
-int aft_get_row_of_hash(HASH_TYPE *hash, struct osl_row **row)
+static int aft_get_row_of_hash(HASH_TYPE *hash, struct osl_row **row)
{
const struct osl_object obj = {.data = hash, .size = HASH_SIZE};
- return osl_get_row(audio_file_table, AFTCOL_HASH, &obj, row);
+ return osl(osl_get_row(audio_file_table, AFTCOL_HASH, &obj, row));
}
/**
* \param row Pointer to a row in the audio file table.
* \param obj Result pointer.
*
- * \return The return value of the underlying call to osl_get_object().
+ * \return Standard.
*/
int get_afsi_object_of_row(const struct osl_row *row, struct osl_object *obj)
{
- return osl_get_object(audio_file_table, row, AFTCOL_AFSI, obj);
+ return osl(osl_get_object(audio_file_table, row, AFTCOL_AFSI, obj));
}
/**
*
* \return Positive on success, negative on errors.
*/
-int get_afsi_object_of_path(const char *path, struct osl_object *obj)
+static int get_afsi_object_of_path(const char *path, struct osl_object *obj)
{
struct osl_row *row;
int ret = aft_get_row_of_path(path, &row);
int get_audio_file_path_of_row(const struct osl_row *row, char **path)
{
struct osl_object path_obj;
- int ret = osl_get_object(audio_file_table, row, AFTCOL_PATH,
- &path_obj);
+ int ret = osl(osl_get_object(audio_file_table, row, AFTCOL_PATH,
+ &path_obj));
if (ret < 0)
return ret;
*path = path_obj.data;
*
* \sa get_hash_of_row().
*/
-static int get_hash_object_of_aft_row(const struct osl_row *row, struct osl_object *obj)
+static int get_hash_object_of_aft_row(const struct osl_row *row,
+ struct osl_object *obj)
{
- return osl_get_object(audio_file_table, row, AFTCOL_HASH, obj);
+ return osl(osl_get_object(audio_file_table, row, AFTCOL_HASH, obj));
}
/**
int get_afhi_of_row(const struct osl_row *row, struct afh_info *afhi)
{
struct osl_object obj;
- int ret = osl_get_object(audio_file_table, row, AFTCOL_AFHI,
- &obj);
+ int ret = osl(osl_get_object(audio_file_table, row, AFTCOL_AFHI,
+ &obj));
if (ret < 0)
return ret;
load_afhi(obj.data, afhi);
ret = shm_attach(shmid, ATTACH_RW, &shm_afd);
if (ret < 0)
goto err;
- *(struct audio_file_data *)shm_afd = *afd;
buf = shm_afd;
buf += sizeof(*afd);
- save_chunk_table(&afd->afhi, buf);
+ afd->max_chunk_size = save_chunk_table(&afd->afhi, buf);
+ *(struct audio_file_data *)shm_afd = *afd;
shm_detach(shm_afd);
return shmid;
err:
return ret;
}
+/**
+ * Extract a afd stored in a shared memory area.
+ *
+ * Attach the shared memory area given by \a shmid, load the audio file data
+ * stored therein and detach the area afterwards. Called by vss, after
+ * receiving a positive response to the request for the next audio file.
+ +
+ * \param shmid The identifier of the shared memory area containing the afd.
+ * \param afd Result pointer.
+ *
+ * \return Standard.
+ */
int load_afd(int shmid, struct audio_file_data *afd)
{
void *shm_afd;
}
}
-static char *make_attribute_lines(const char *att_bitmap, struct afs_info *afsi)
+
+static int write_attribute_items(struct para_buffer *b,
+ const char *att_bitmap, struct afs_info *afsi)
{
- char *att_text, *att_lines;
+ char *att_text;
+ int ret;
- get_attribute_text(&afsi->attributes, " ", &att_text);
- if (!att_text)
- return para_strdup(att_bitmap);
- att_lines = make_message("%s: %s\n%s: %s",
- status_item_list[SI_ATTRIBUTES_BITMAP], att_bitmap,
- status_item_list[SI_ATTRIBUTES_TXT], att_text);
+ ret = WRITE_STATUS_ITEM(b, SI_ATTRIBUTES_BITMAP, "%s\n", att_bitmap);
+ if (ret < 0)
+ return ret;
+ ret = get_attribute_text(&afsi->attributes, " ", &att_text);
+ if (ret < 0)
+ return ret;
+ ret = WRITE_STATUS_ITEM(b, SI_ATTRIBUTES_TXT, "%s\n", att_text);
free(att_text);
- return att_lines;
+ return ret;
}
-static char *make_lyrics_lines(struct afs_info *afsi)
+static int write_lyrics_items(struct para_buffer *b, struct afs_info *afsi)
{
char *lyrics_name;
+ int ret;
+ ret = WRITE_STATUS_ITEM(b, SI_LYRICS_ID, "%u\n", afsi->lyrics_id);
+ if (ret < 0)
+ return ret;
lyr_get_name_by_id(afsi->lyrics_id, &lyrics_name);
- return make_message("%s: %u\n%s: %s\n",
- status_item_list[SI_LYRICS_ID], afsi->lyrics_id,
- status_item_list[SI_LYRICS_NAME], lyrics_name?
- lyrics_name : "(none)");
+ return WRITE_STATUS_ITEM(b, SI_LYRICS_NAME, "%s\n", lyrics_name?
+ lyrics_name : "(none)");
}
-static char *make_image_lines(struct afs_info *afsi)
+static int write_image_items(struct para_buffer *b, struct afs_info *afsi)
{
char *image_name;
+ int ret;
+
+ ret = WRITE_STATUS_ITEM(b, SI_IMAGE_ID, "%u\n", afsi->image_id);
+ if (ret < 0)
+ return ret;
img_get_name_by_id(afsi->image_id, &image_name);
- return make_message("%s: %u\n%s: %s\n",
- status_item_list[SI_IMAGE_ID], afsi->image_id,
- status_item_list[SI_IMAGE_NAME], image_name?
- image_name : "(none)");
+ return WRITE_STATUS_ITEM(b, SI_IMAGE_NAME, "%s\n", image_name?
+ image_name : "(none)");
}
-static char *make_filename_lines(const char *path, unsigned flags)
+static int write_filename_items(struct para_buffer *b, const char *path,
+ unsigned flags)
{
- char *dirname, *ret;
- const char *basename;
+ char *val;
+ int ret;
if (!(flags & LS_FLAG_FULL_PATH))
- return make_message("%s: %s\n",
- status_item_list[SI_BASENAME], path);
- basename = para_basename(path),
- dirname = para_dirname(path);
- ret = make_message("%s: %s\n%s: %s\n%s: %s\n",
- status_item_list[SI_PATH], path,
- status_item_list[SI_DIRECTORY], dirname? dirname : "?",
- status_item_list[SI_BASENAME], basename? basename : "?");
- free(dirname);
+ return WRITE_STATUS_ITEM(b, SI_BASENAME, "%s\n", path);
+ ret = WRITE_STATUS_ITEM(b, SI_PATH, "%s\n", path);
+ if (ret < 0)
+ return ret;
+ val = para_basename(path);
+ ret = WRITE_STATUS_ITEM(b, SI_BASENAME, "%s\n", val? val : "");
+ if (ret < 0)
+ return ret;
+ val = para_dirname(path);
+ ret = WRITE_STATUS_ITEM(b, SI_DIRECTORY, "%s\n", val? val : "");
+ free(val);
return ret;
}
return ret;
}
+static int write_score(struct para_buffer *b, struct ls_data *d,
+ struct ls_options *opts)
+{
+ if (!(opts->flags & LS_FLAG_ADMISSIBLE_ONLY)) /* no score*/
+ return 0;
+ return WRITE_STATUS_ITEM(b, SI_SCORE, "%li\n", d->score);
+}
+
static int print_list_item(struct ls_data *d, struct ls_options *opts,
struct para_buffer *b, time_t current_time)
{
char att_buf[65];
char last_played_time[30];
char duration_buf[30]; /* nobody has an audio file long enough to overflow this */
- char score_buf[30] = "";
struct afs_info *afsi = &d->afsi;
struct afh_info *afhi = &d->afhi;
- struct ls_widths *w = &opts->widths;
- int have_score = opts->flags & LS_FLAG_ADMISSIBLE_ONLY;
char asc_hash[2 * HASH_SIZE + 1];
- char *att_lines, *lyrics_lines, *image_lines, *filename_lines;
if (opts->mode == LS_MODE_SHORT) {
ret = para_printf(b, "%s\n", d->path);
goto out;
}
get_duration_buf(afhi->seconds_total, duration_buf, opts);
- if (have_score) {
- if (opts->mode == LS_MODE_LONG)
- sprintf(score_buf, "%*li ", w->score_width, d->score);
- else
- sprintf(score_buf, "%li ", d->score);
- }
-
if (opts->mode == LS_MODE_LONG) {
+ struct ls_widths *w = &opts->widths;
+ if (opts->flags & LS_FLAG_ADMISSIBLE_ONLY) {
+ ret = para_printf(b, "%*li ",
+ opts->widths.score_width, d->score);
+ if (ret < 0)
+ goto out;
+ }
ret = para_printf(b,
- "%s" /* score */
"%s " /* attributes */
"%*u " /* amp */
"%*d " /* image_id */
"%*d " /* num_played */
"%s " /* last_played */
"%s\n", /* path */
- score_buf,
att_buf,
w->amp_width, afsi->amp,
w->image_id_width, afsi->image_id,
);
goto out;
}
- hash_to_asc(d->hash, asc_hash);
- att_lines = make_attribute_lines(att_buf, afsi);
- lyrics_lines = make_lyrics_lines(afsi);
- image_lines = make_image_lines(afsi);
- filename_lines = make_filename_lines(d->path, opts->flags);
if (opts->mode == LS_MODE_MBOX) {
const char *bn = para_basename(d->path);
ret = para_printf(b,
if (ret < 0)
goto out;
}
- ret = para_printf(b,
- "%s" /* filename stuff */
- "%s%s%s%s" /* score */
- "%s\n" /* attributes */
- "%s: %s\n" /* hash */
- "%s" /* image id, image name */
- "%s" /* lyrics */
- "%s: %dkbit/s\n" /* bitrate */
- "%s: %s\n" /* format */
- "%s: %dHz\n" /* frequency */
- "%s: %d\n" /* channels */
- "%s: %s\n" /* duration */
- "%s: %lu\n" /* seconds total */
- "%s: %s\n" /* last played time */
- "%s: %d\n" /* num_played */
- "%s: %u\n" /* ampplification */
- "%s" /* tag info */
- "%s: %lu\n" /* chunk time */
- "%s: %lu\n", /* num chunks */
- filename_lines,
- have_score? status_item_list[SI_SCORE] : "",
- have_score? ": " : "",
- score_buf,
- have_score? "\n" : "",
- att_lines,
- status_item_list[SI_HASH], asc_hash,
- image_lines,
- lyrics_lines,
- status_item_list[SI_BITRATE], afhi->bitrate,
- status_item_list[SI_FORMAT], audio_format_name(afsi->audio_format_id),
- status_item_list[SI_FREQUENCY], afhi->frequency,
- status_item_list[SI_CHANNELS], afhi->channels,
- status_item_list[SI_DURATION], duration_buf,
- status_item_list[SI_SECONDS_TOTAL], afhi->seconds_total,
- status_item_list[SI_LAST_PLAYED], last_played_time,
- status_item_list[SI_NUM_PLAYED], afsi->num_played,
- status_item_list[SI_AMPLIFICATION], afsi->amp,
- afhi->info_string,
- status_item_list[SI_CHUNK_TIME], tv2ms(&afhi->chunk_tv),
- status_item_list[SI_NUM_CHUNKS], afhi->chunks_total
- );
+ ret = write_filename_items(b, d->path, opts->flags);
+ if (ret < 0)
+ goto out;
+ ret = write_score(b, d, opts);
+ if (ret < 0)
+ goto out;
+ ret = write_attribute_items(b, att_buf, afsi);
+ if (ret < 0)
+ goto out;
+ ret = write_image_items(b, afsi);
+ if (ret < 0)
+ goto out;
+ ret = write_lyrics_items(b, afsi);
+ if (ret < 0)
+ goto out;
+ hash_to_asc(d->hash, asc_hash);
+ ret = WRITE_STATUS_ITEM(b, SI_HASH, "%s\n", asc_hash);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_BITRATE, "%dkbit/s\n", afhi->bitrate);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_FORMAT, "%s\n",
+ audio_format_name(afsi->audio_format_id));
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_FREQUENCY, "%dHz\n", afhi->frequency);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_CHANNELS, "%d\n", afhi->channels);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_DURATION, "%s\n", duration_buf);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_SECONDS_TOTAL, "%lu\n",
+ afhi->seconds_total);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_LAST_PLAYED, "%s\n", last_played_time);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_NUM_PLAYED, "%d\n", afsi->num_played);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_AMPLIFICATION, "%u\n", afsi->amp);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_CHUNK_TIME, "%lu\n",
+ tv2ms(&afhi->chunk_tv));
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_NUM_CHUNKS, "%lu\n",
+ afhi->chunks_total);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_TECHINFO, "%s\n", afhi->techinfo);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_ARTIST, "%s\n", afhi->tags.artist);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_TITLE, "%s\n", afhi->tags.title);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_YEAR, "%s\n", afhi->tags.year);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_ALBUM, "%s\n", afhi->tags.album);
+ if (ret < 0)
+ goto out;
+ ret = WRITE_STATUS_ITEM(b, SI_COMMENT, "%s\n", afhi->tags.comment);
if (ret < 0)
goto out;
if (opts->mode == LS_MODE_MBOX) {
osl_close_disk_object(&lyrics_def);
}
}
- free(att_lines);
- free(lyrics_lines);
- free(image_lines);
- free(filename_lines);
out:
- free(afhi->info_string);
return ret;
}
-/**
- * Write a list of audio-file related status items with empty values.
- *
- * \param buf Result pointer.
- *
- * This is used by vss when currently no audio file is open.
- */
-void make_empty_status_items(char *buf)
-{
- sprintf(buf,
- "%s: \n" /* path */
- "%s: \n" /* dirname */
- "%s: \n" /* basename */
- "%s: \n" /* score */
- "%s: \n" /* attributes bitmap */
- "%s: \n" /* attributes txt */
- "%s: \n" /* hash */
- "%s: \n" /* image id */
- "%s: \n" /* image name */
- "%s: \n" /* lyrics id */
- "%s: \n" /* lyrics name */
- "%s: \n" /* bitrate */
- "%s: \n" /* format */
- "%s: \n" /* frequency */
- "%s: \n" /* channels */
- "%s: \n" /* duration */
- "%s: \n" /* seconds total */
- "%s: \n" /* num played */
- "%s: \n" /* last played */
- "%s: \n" /* audio file info */
- "%s: \n" /* taginfo1 */
- "%s: \n" /* taginfo2 */
- "%s: \n" /* amplification */
- ,
- status_item_list[SI_PATH],
- status_item_list[SI_DIRECTORY],
- status_item_list[SI_BASENAME],
- status_item_list[SI_SCORE],
- status_item_list[SI_ATTRIBUTES_BITMAP],
- status_item_list[SI_ATTRIBUTES_TXT],
- status_item_list[SI_HASH],
- status_item_list[SI_IMAGE_ID],
- status_item_list[SI_IMAGE_NAME],
- status_item_list[SI_LYRICS_ID],
- status_item_list[SI_LYRICS_NAME],
- status_item_list[SI_BITRATE],
- status_item_list[SI_FORMAT],
- status_item_list[SI_FREQUENCY],
- status_item_list[SI_CHANNELS],
- status_item_list[SI_DURATION],
- status_item_list[SI_SECONDS_TOTAL],
- status_item_list[SI_NUM_PLAYED],
- status_item_list[SI_LAST_PLAYED],
- status_item_list[SI_AUDIO_FILE_INFO],
- status_item_list[SI_TAGINFO1],
- status_item_list[SI_TAGINFO2],
- status_item_list[SI_AMPLIFICATION]
- );
-}
-
-static void fixup_taginfo(char *begin, char *end)
-{
- char *p = begin;
-
- for (;;) {
- p = strchr(p, '\n');
- if (!p)
- break;
- if (p >= end - 1)
- break;
- *p = ' ';
- p++;
- }
-}
-
-/* crap, remove this ASAP. */
-static int fixup_info_string(char *info_string)
-{
- char *t1, *t2, *end;
-
- if (strncmp(info_string, "audio_file_info:", 16))
- return -ERRNO_TO_PARA_ERROR(EINVAL);
- t1 = strstr(info_string, "\ntaginfo1:");
- if (!t1)
- return -ERRNO_TO_PARA_ERROR(EINVAL);
- t2 = strstr(info_string, "\ntaginfo2: ");
- if (!t2)
- return -ERRNO_TO_PARA_ERROR(EINVAL);
-
- end = t2 + strlen(t2);
- fixup_taginfo(info_string + 16, t1);
- fixup_taginfo(t1 + 10, t2);
- fixup_taginfo(t2 + 10, end);
-
- if (t1 - info_string < 80 && t2 - t1 < 80 && end - t2 < 80)
- return 0;
- if (t1 - info_string >= 80) {
- memmove(info_string + 80, t1, end - t1);
- t1 = info_string + 80;
- t2 -= t1 - info_string - 80;
- end -= t1 - info_string - 80;
- }
- if (t2 - t1 >= 80) {
- memmove(t1 + 80, t2, end - t2);
- end -= t2 - t1 - 80;
- t2 = t1 + 80;
- }
- if (end - t2 >= 80) {
- t2[78] = '\n';
- t2[79] = '\0';
- }
- return 1;
-}
-
static int make_status_items(struct audio_file_data *afd,
struct afs_info *afsi, char *path, long score,
HASH_TYPE *hash)
.flags = LS_FLAG_FULL_PATH | LS_FLAG_ADMISSIBLE_ONLY,
.mode = LS_MODE_VERBOSE,
};
- struct para_buffer pb = {.max_size = VERBOSE_LS_OUTPUT_SIZE - 1};
+ struct para_buffer pb = {.max_size = SHMMAX - 1};
time_t current_time;
int ret;
- ret = fixup_info_string(afd->afhi.info_string);
- if (ret < 0) {
- PARA_WARNING_LOG("ignoring invalid tag info\n");
- afd->afhi.info_string[0] = '\0';
- } else if (ret)
- PARA_NOTICE_LOG("truncated overlong tag info\n");
time(¤t_time);
- ret = print_list_item(&d, &opts, &pb, current_time); /* frees info string */
- afd->afhi.info_string = NULL;
+ ret = print_list_item(&d, &opts, &pb, current_time);
if (ret < 0)
- goto out;
- strncpy(afd->verbose_ls_output, pb.buf, VERBOSE_LS_OUTPUT_SIZE);
- afd->verbose_ls_output[VERBOSE_LS_OUTPUT_SIZE - 1] = '\0';
-out:
- free(pb.buf);
- return ret;
+ return ret;
+ free(status_items);
+ status_items = pb.buf;
+ memset(&pb, 0, sizeof(pb));
+ pb.max_size = SHMMAX - 1;
+ pb.flags = PBF_SIZE_PREFIX;
+ ret = print_list_item(&d, &opts, &pb, current_time);
+ if (ret < 0) {
+ free(status_items);
+ status_items = NULL;
+ return ret;
+ }
+ free(parser_friendly_status_items);
+ parser_friendly_status_items = pb.buf;
+ return 1;
}
/**
ret = get_audio_file_path_of_row(aft_row, &path);
if (ret < 0)
return ret;
+ PARA_NOTICE_LOG("%s\n", path);
ret = get_afsi_object_of_row(aft_row, &afsi_obj);
if (ret < 0)
return ret;
ret = save_afd(afd);
err:
free(afd->afhi.chunk_table);
- free(afd->afhi.info_string);
osl_close_disk_object(&chunk_table_obj);
return ret;
}
}
return 1;
err:
- free(d->afhi.info_string);
return ret;
}
struct ls_options *opts = query->data;
char *p, *pattern_start = (char *)query->data + sizeof(*opts);
struct para_buffer b = {.max_size = SHMMAX,
+ .flags = (opts->mode == LS_MODE_PARSER)? PBF_SIZE_PREFIX : 0,
.max_size_handler = pass_buffer_as_shm, .private_data = &fd};
int i = 0, ret;
time_t current_time;
-
if (opts->num_patterns) {
opts->patterns = para_malloc(opts->num_patterns * sizeof(char *));
for (i = 0, p = pattern_start; i < opts->num_patterns; i++) {
if (opts->flags & LS_FLAG_ADMISSIBLE_ONLY)
ret = admissible_file_loop(opts, prepare_ls_row);
else
- ret = osl_rbtree_loop(audio_file_table, AFTCOL_PATH, opts,
- prepare_ls_row);
+ ret = osl(osl_rbtree_loop(audio_file_table, AFTCOL_PATH, opts,
+ prepare_ls_row));
if (ret < 0)
goto out;
if (!opts->num_matching_paths)
/*
* TODO: flags -h (sort by hash)
*/
-int com_ls(int fd, int argc, char * const * const argv)
+int com_ls(struct rc4_context *rc4c, int argc, char * const * const argv)
{
int i, ret;
unsigned flags = 0;
case 'c':
mode = LS_MODE_CHUNKS;
continue;
+ case 'p':
+ mode = LS_MODE_PARSER;
+ continue;
default:
return -E_AFT_SYNTAX;
}
opts.mode = mode;
opts.num_patterns = argc - i;
ret = send_option_arg_callback_request(&query, opts.num_patterns,
- argv + i, com_ls_callback, send_result, &fd);
+ argv + i, com_ls_callback, rc4_send_result, rc4c);
return ret;
}
* \param private_data An arbitrary data pointer, passed to \a func.
* \param func The custom function to be called.
*
- * \return The return value of the underlying call to osl_rbtree_loop().
+ * \return Standard.
*/
int audio_file_loop(void *private_data, osl_rbtree_loop_func *func)
{
- return osl_rbtree_loop(audio_file_table, AFTCOL_HASH, private_data,
- func);
+ return osl(osl_rbtree_loop(audio_file_table, AFTCOL_HASH, private_data,
+ func));
}
static struct osl_row *find_hash_sister(HASH_TYPE *hash)
PARA_INFO_LOG("request to add %s\n", path);
hs = find_hash_sister(hash);
ret = aft_get_row_of_path(path, &pb);
- if (ret < 0 && ret != -E_RB_KEY_NOT_FOUND)
+ if (ret < 0 && ret != -OSL_ERRNO_TO_PARA_ERROR(E_OSL_RB_KEY_NOT_FOUND))
goto out;
if (hs && pb && hs == pb && !(flags & ADD_FLAG_FORCE)) {
if (flags & ADD_FLAG_VERBOSE)
if (ret < 0)
goto out;
}
- ret = osl_del_row(audio_file_table, pb);
+ ret = osl(osl_del_row(audio_file_table, pb));
if (ret < 0)
goto out;
pb = NULL;
}
/* file rename, update hs' path */
if (flags & ADD_FLAG_VERBOSE) {
- ret = osl_get_object(audio_file_table, hs,
- AFTCOL_PATH, &obj);
+ ret = osl(osl_get_object(audio_file_table, hs,
+ AFTCOL_PATH, &obj));
if (ret < 0)
goto out;
ret = para_printf(&msg, "renamed from %s\n", (char *)obj.data);
if (ret < 0)
goto out;
}
- ret = osl_update_object(audio_file_table, hs, AFTCOL_PATH,
- &objs[AFTCOL_PATH]);
+ ret = osl(osl_update_object(audio_file_table, hs, AFTCOL_PATH,
+ &objs[AFTCOL_PATH]));
if (ret < 0)
goto out;
afs_event(AUDIO_FILE_RENAME, &msg, hs);
if (ret < 0)
goto out;
}
- ret = osl_update_object(audio_file_table, row, AFTCOL_AFHI,
- &objs[AFTCOL_AFHI]);
+ ret = osl(osl_update_object(audio_file_table, row, AFTCOL_AFHI,
+ &objs[AFTCOL_AFHI]));
if (ret < 0)
goto out;
- ret = osl_update_object(audio_file_table, row, AFTCOL_CHUNKS,
- &objs[AFTCOL_CHUNKS]);
+ ret = osl(osl_update_object(audio_file_table, row, AFTCOL_CHUNKS,
+ &objs[AFTCOL_CHUNKS]));
if (ret < 0)
goto out;
afs_event(AFHI_CHANGE, &msg, row);
objs[AFTCOL_AFSI].data = &afsi_buf;
objs[AFTCOL_AFSI].size = AFSI_SIZE;
save_afsi(&default_afsi, &objs[AFTCOL_AFSI]);
- ret = osl_add_and_get_row(audio_file_table, objs, &aft_row);
+ ret = osl(osl_add_and_get_row(audio_file_table, objs, &aft_row));
afs_event(AUDIO_FILE_ADD, &msg, aft_row);
out:
if (ret < 0)
/** Used by com_add(). */
struct private_add_data {
- /** The socket file descriptor. */
- int fd;
+ /** The socket file descriptor, including rc4 keys. */
+ struct rc4_context *rc4c;
/** The given add flags. */
uint32_t flags;
};
query.size = strlen(path) + 1;
ret = send_callback_request(path_brother_callback, &query,
get_row_pointer_from_result, &pb);
- if (ret < 0 && ret != -E_RB_KEY_NOT_FOUND)
+ if (ret < 0 && ret != -OSL_ERRNO_TO_PARA_ERROR(E_OSL_RB_KEY_NOT_FOUND))
goto out_free;
ret = 1;
if (pb && (pad->flags & ADD_FLAG_LAZY)) { /* lazy is really cheap */
if (pad->flags & ADD_FLAG_VERBOSE)
- send_ret = send_va_buffer(pad->fd, "lazy-ignore: %s\n", path);
+ send_ret = rc4_send_va_buffer(pad->rc4c,
+ "lazy-ignore: %s\n", path);
goto out_free;
}
/* We still want to add this file. Compute its hash. */
ret = 1;
if (pb && hs && hs == pb && !(pad->flags & ADD_FLAG_FORCE)) {
if (pad->flags & ADD_FLAG_VERBOSE)
- send_ret = send_va_buffer(pad->fd,
+ send_ret = rc4_send_va_buffer(pad->rc4c,
"%s exists, not forcing update\n", path);
goto out_unmap;
}
munmap(map.data, map.size);
close(fd);
if (pad->flags & ADD_FLAG_VERBOSE) {
- send_ret = send_va_buffer(pad->fd, "adding %s\n", path);
+ send_ret = rc4_send_va_buffer(pad->rc4c, "adding %s\n", path);
if (send_ret < 0)
goto out_free;
}
save_add_callback_buffer(hash, path, afhi_ptr, pad->flags, format_num, &obj);
/* Ask afs to consider this entry for adding. */
- ret = send_callback_request(com_add_callback, &obj, send_result, &pad->fd);
+ ret = send_callback_request(com_add_callback, &obj, rc4_send_result, pad->rc4c);
goto out_free;
out_unmap:
munmap(map.data, map.size);
out_free:
if (ret < 0 && send_ret >= 0)
- send_ret = send_va_buffer(pad->fd, "failed to add %s (%s)\n", path,
- para_strerror(-ret));
+ send_ret = rc4_send_va_buffer(pad->rc4c,
+ "failed to add %s (%s)\n", path, para_strerror(-ret));
free(obj.data);
if (afhi_ptr) {
free(afhi_ptr->chunk_table);
- free(afhi_ptr->info_string);
+ free(afhi_ptr->techinfo);
+ free(afhi_ptr->tags.artist);
+ free(afhi_ptr->tags.title);
+ free(afhi_ptr->tags.year);
+ free(afhi_ptr->tags.album);
+ free(afhi_ptr->tags.comment);
}
/* Stop adding files only on send errors. */
return send_ret;
}
-int com_add(int fd, int argc, char * const * const argv)
+int com_add(struct rc4_context *rc4c, int argc, char * const * const argv)
{
int i, ret;
- struct private_add_data pad = {.fd = fd, .flags = 0};
+ struct private_add_data pad = {.rc4c = rc4c, .flags = 0};
struct stat statbuf;
for (i = 1; i < argc; i++) {
char *path;
ret = verify_path(argv[i], &path);
if (ret < 0) {
- ret = send_va_buffer(fd, "%s: %s\n", argv[i],
+ ret = rc4_send_va_buffer(rc4c, "%s: %s\n", argv[i],
para_strerror(-ret));
if (ret < 0)
return ret;
}
ret = stat(path, &statbuf);
if (ret < 0) {
- ret = send_va_buffer(fd, "failed to stat %s (%s)\n", path,
+ ret = rc4_send_va_buffer(rc4c, "failed to stat %s (%s)\n", path,
strerror(errno));
free(path);
if (ret < 0)
else
ret = add_one_audio_file(path, &pad);
if (ret < 0) {
- send_va_buffer(fd, "%s: %s\n", path, para_strerror(-ret));
+ rc4_send_va_buffer(rc4c, "%s: %s\n", path, para_strerror(-ret));
free(path);
return ret;
}
free(tad.pb.buf);
}
-int com_touch(int fd, int argc, char * const * const argv)
+int com_touch(struct rc4_context *rc4c, int argc, char * const * const argv)
{
struct com_touch_options cto = {
.num_played = -1,
i++;
break;
}
- if (!strncmp(arg, "-n", 2)) {
- ret = para_atoi32(arg + 2, &cto.num_played);
+ if (!strncmp(arg, "-n=", 3)) {
+ ret = para_atoi32(arg + 3, &cto.num_played);
if (ret < 0)
return ret;
continue;
}
- if (!strncmp(arg, "-l", 2)) {
- ret = para_atoi64(arg + 2, &cto.last_played);
+ if (!strncmp(arg, "-l=", 3)) {
+ ret = para_atoi64(arg + 3, &cto.last_played);
if (ret < 0)
return ret;
continue;
}
- if (!strncmp(arg, "-y", 2)) {
- ret = para_atoi32(arg + 2, &cto.lyrics_id);
+ if (!strncmp(arg, "-y=", 3)) {
+ ret = para_atoi32(arg + 3, &cto.lyrics_id);
if (ret < 0)
return ret;
continue;
}
- if (!strncmp(arg, "-i", 2)) {
- ret = para_atoi32(arg + 2, &cto.image_id);
+ if (!strncmp(arg, "-i=", 3)) {
+ ret = para_atoi32(arg + 3, &cto.image_id);
if (ret < 0)
return ret;
continue;
}
- if (!strncmp(arg, "-a", 2)) {
+ if (!strncmp(arg, "-a=", 3)) {
int32_t val;
- ret = para_atoi32(arg + 2, &val);
+ ret = para_atoi32(arg + 3, &val);
if (ret < 0)
return ret;
if (val < 0 || val > 255)
if (i >= argc)
return -E_AFT_SYNTAX;
ret = send_option_arg_callback_request(&query, argc - i,
- argv + i, com_touch_callback, send_result, &fd);
+ argv + i, com_touch_callback, rc4_send_result, rc4c);
if (ret < 0)
- send_va_buffer(fd, "%s\n", para_strerror(-ret));
+ rc4_send_va_buffer(rc4c, "%s\n", para_strerror(-ret));
return ret;
}
return ret;
}
afs_event(AUDIO_FILE_REMOVE, &crd->pb, row);
- ret = osl_del_row(audio_file_table, row);
+ ret = osl(osl_del_row(audio_file_table, row));
if (ret < 0)
para_printf(&crd->pb, "%s: %s\n", name, para_strerror(-ret));
else
}
/* TODO options: -r (recursive) */
-int com_rm(int fd, int argc, char * const * const argv)
+int com_rm(struct rc4_context *rc4c, int argc, char * const * const argv)
{
uint32_t flags = 0;
struct osl_object query = {.data = &flags, .size = sizeof(flags)};
if (i >= argc)
return -E_AFT_SYNTAX;
ret = send_option_arg_callback_request(&query, argc - i, argv + i,
- com_rm_callback, send_result, &fd);
+ com_rm_callback, rc4_send_result, rc4c);
if (ret < 0)
- send_va_buffer(fd, "%s\n", para_strerror(-ret));
+ rc4_send_va_buffer(rc4c, "%s\n", para_strerror(-ret));
return ret;
}
free(cad.pb.buf);
}
-int com_cpsi(int fd, int argc, char * const * const argv)
+int com_cpsi(struct rc4_context *rc4c, int argc, char * const * const argv)
{
unsigned flags = 0;
int i, ret;
}
break;
}
- if (i + 1 >= argc) /* need at least souce file and pattern */
+ if (i + 1 >= argc) /* need at least source file and pattern */
return -E_AFT_SYNTAX;
if (!(flags & ~CPSI_FLAG_VERBOSE)) /* no copy flags given */
flags = ~(unsigned)CPSI_FLAG_VERBOSE | flags;
ret = send_option_arg_callback_request(&options, argc - i, argv + i,
- com_cpsi_callback, send_result, &fd);
+ com_cpsi_callback, rc4_send_result, rc4c);
if (ret < 0)
- send_va_buffer(fd, "%s\n", para_strerror(-ret));
+ rc4_send_va_buffer(rc4c, "%s\n", para_strerror(-ret));
return ret;
}
+static void afs_stat_callback(int fd, const struct osl_object *query)
+{
+ int *parser_friendly = query->data;
+ char *buf = *parser_friendly?
+ parser_friendly_status_items : status_items;
+
+ if (!buf)
+ return;
+ pass_buffer_as_shm(buf, strlen(buf), &fd);
+}
+
+/**
+ * Get the current afs status items from the afs process and send it using RC4.
+ *
+ * \param rc4c The rc4 context for data encryption.
+ * \param parser_friendly Whether parser-friendly output format should be used.
+ *
+ * As the contents of the afs status items change in time and the command
+ * handler only has a COW version created at fork time, it can not send
+ * up-to-date afs status items directly. Therefore the usual callback mechanism
+ * is used to pass the status items from the afs process to the command handler
+ * via a shared memory area and a pipe.
+ */
+int send_afs_status(struct rc4_context *rc4c, int parser_friendly)
+{
+ struct osl_object query = {.data = &parser_friendly,
+ .size = sizeof(parser_friendly)};
+
+ return send_callback_request(afs_stat_callback, &query, rc4_send_result, rc4c);
+}
+
/* TODO: optionally fix problems by removing offending rows */
static int check_audio_file(struct osl_row *row, void *data)
{
{
osl_close_table(audio_file_table, OSL_MARK_CLEAN);
audio_file_table = NULL;
+ free(status_items);
+ status_items = NULL;
+ free(parser_friendly_status_items);
+ parser_friendly_status_items = NULL;
}
/**
int ret;
audio_file_table_desc.dir = dir;
- ret = osl_open_table(&audio_file_table_desc, &audio_file_table);
+ ret = osl(osl_open_table(&audio_file_table_desc, &audio_file_table));
if (ret >= 0) {
unsigned num;
osl_get_num_rows(audio_file_table, &num);
}
PARA_INFO_LOG("failed to open audio file table\n");
audio_file_table = NULL;
- if (ret >= 0 || is_errno(-ret, ENOENT))
+ if (ret >= 0 || ret == -OSL_ERRNO_TO_PARA_ERROR(E_OSL_NOENT))
return 1;
return ret;
}
static int aft_create(const char *dir)
{
audio_file_table_desc.dir = dir;
- return osl_create_table(&audio_file_table_desc);
+ return osl(osl_create_table(&audio_file_table_desc));
}
static int clear_attribute(struct osl_row *row, void *data)
}
}
+/**
+ * Initialize the audio file table.
+ *
+ * \param t Pointer to the structure to be initialized.
+ */
void aft_init(struct afs_table *t)
{
t->open = aft_open;
/*
- * Copyright (C) 2005-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2005-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
* based on the vplay program by Michael Beck.
*/
+#include <regex.h>
#include <sys/types.h>
#include <dirent.h>
#include <alsa/asoundlib.h>
+#include <sys/time.h>
+#include <stdbool.h>
#include "para.h"
#include "fd.h"
#include "list.h"
#include "sched.h"
#include "ggo.h"
+#include "buffer_tree.h"
#include "write.h"
+#include "write_common.h"
#include "alsa_write.cmdline.h"
#include "error.h"
-/** always use 16 bit little endian */
-#define FORMAT SND_PCM_FORMAT_S16_LE
-
/** Data specific to the alsa writer. */
struct private_alsa_write_data {
/** The alsa handle */
snd_pcm_t *handle;
- /** Determined and set by alsa_open(). */
+ /** Determined and set by alsa_init(). */
int bytes_per_frame;
/** The approximate maximum buffer duration in us. */
unsigned buffer_time;
/* Number of frames that fit into the buffer. */
- unsigned buffer_frames;
+ snd_pcm_uframes_t buffer_frames;
/**
- * The samplerate given by command line option or the decoder
+ * The sample rate given by command line option or the decoder
* of the writer node group.
*/
- unsigned samplerate;
+ unsigned sample_rate;
+ snd_pcm_format_t sample_format;
/**
* The number of channels, given by command line option or the
* decoder of the writer node group.
*/
unsigned channels;
+ struct timeval drain_barrier;
};
+static snd_pcm_format_t get_alsa_pcm_format(enum sample_format sf)
+{
+ switch (sf) {
+ case SF_S8: return SND_PCM_FORMAT_S8;
+ case SF_U8: return SND_PCM_FORMAT_U8;
+ case SF_S16_LE: return SND_PCM_FORMAT_S16_LE;
+ case SF_S16_BE: return SND_PCM_FORMAT_S16_BE;
+ case SF_U16_LE: return SND_PCM_FORMAT_U16_LE;
+ case SF_U16_BE: return SND_PCM_FORMAT_U16_BE;
+ default: return SND_PCM_FORMAT_S16_LE;
+ }
+}
+
/* Install PCM software and hardware configuration. */
static int alsa_init(struct private_alsa_write_data *pad,
struct alsa_write_args_info *conf)
{
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
- snd_pcm_uframes_t buffer_size, start_threshold, stop_threshold;
+ snd_pcm_uframes_t start_threshold, stop_threshold;
snd_pcm_uframes_t period_size;
int err;
+ PARA_INFO_LOG("opening %s\n", conf->device_arg);
err = snd_pcm_open(&pad->handle, conf->device_arg,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (err < 0)
if (snd_pcm_hw_params_set_access(pad->handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED) < 0)
return -E_ACCESS_TYPE;
- if (snd_pcm_hw_params_set_format(pad->handle, hwparams, FORMAT) < 0)
+ if (snd_pcm_hw_params_set_format(pad->handle, hwparams,
+ pad->sample_format) < 0)
return -E_SAMPLE_FORMAT;
if (snd_pcm_hw_params_set_channels(pad->handle, hwparams,
pad->channels) < 0)
return -E_CHANNEL_COUNT;
if (snd_pcm_hw_params_set_rate_near(pad->handle, hwparams,
- &pad->samplerate, NULL) < 0)
+ &pad->sample_rate, NULL) < 0)
return -E_SET_RATE;
err = snd_pcm_hw_params_get_buffer_time_max(hwparams,
&pad->buffer_time, NULL);
if (snd_pcm_hw_params(pad->handle, hwparams) < 0)
return -E_HW_PARAMS;
snd_pcm_hw_params_get_period_size(hwparams, &period_size, NULL);
- snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
- PARA_INFO_LOG("buffer size: %lu, period_size: %lu\n", buffer_size,
+ snd_pcm_hw_params_get_buffer_size(hwparams, &pad->buffer_frames);
+ PARA_INFO_LOG("buffer size: %lu, period_size: %lu\n", pad->buffer_frames,
period_size);
- if (period_size == buffer_size)
+ if (period_size == pad->buffer_frames)
return -E_BAD_PERIOD;
snd_pcm_sw_params_current(pad->handle, swparams);
snd_pcm_sw_params_set_avail_min(pad->handle, swparams, period_size);
- if (buffer_size < 1)
+ if (pad->buffer_frames < 1)
start_threshold = 1;
else
- start_threshold = PARA_MIN(buffer_size,
- (snd_pcm_uframes_t)pad->samplerate);
+ start_threshold = PARA_MIN(pad->buffer_frames,
+ (snd_pcm_uframes_t)pad->sample_rate);
if (snd_pcm_sw_params_set_start_threshold(pad->handle, swparams,
start_threshold) < 0)
return -E_START_THRESHOLD;
- stop_threshold = buffer_size;
+ stop_threshold = pad->buffer_frames;
if (snd_pcm_sw_params_set_stop_threshold(pad->handle, swparams,
stop_threshold) < 0)
return -E_STOP_THRESHOLD;
if (snd_pcm_sw_params(pad->handle, swparams) < 0)
PARA_WARNING_LOG("unable to install sw params\n");
- pad->bytes_per_frame = snd_pcm_format_physical_width(FORMAT)
+ pad->bytes_per_frame = snd_pcm_format_physical_width(pad->sample_format)
* pad->channels / 8;
if (pad->bytes_per_frame <= 0)
return -E_PHYSICAL_WIDTH;
PARA_INFO_LOG("bytes per frame: %d\n", pad->bytes_per_frame);
if (snd_pcm_nonblock(pad->handle, 1))
PARA_ERROR_LOG("failed to set nonblock mode\n");
- pad->buffer_frames = 1000 * pad->buffer_time / pad->samplerate;
- PARA_INFO_LOG("max buffered frames: %d\n", pad->buffer_frames);
return 1;
}
-/* Open an instance of the alsa writer. */
-static int alsa_open(struct writer_node *wn)
-{
- struct alsa_write_args_info *conf = wn->conf;
- struct writer_node_group *wng = wn->wng;
- struct private_alsa_write_data *pad = para_calloc(sizeof(*pad));
-
- wn->private_data = pad;
- if (!conf->samplerate_given && wng->samplerate)
- pad->samplerate = *wng->samplerate;
- else
- pad->samplerate = conf->samplerate_arg;
- if (!conf->channels_given && wng->channels)
- pad->channels = *wng->channels;
- else
- pad->channels = conf->channels_arg;
- PARA_INFO_LOG("%d channel(s), %dHz\n", pad->channels, pad->samplerate);
- return 1;
-}
-
-static int alsa_write_pre_select(struct sched *s, struct writer_node *wn)
+static void alsa_write_pre_select(struct sched *s, struct task *t)
{
+ struct writer_node *wn = container_of(t, struct writer_node, task);
struct private_alsa_write_data *pad = wn->private_data;
- struct writer_node_group *wng = wn->wng;
struct timeval tv;
snd_pcm_sframes_t avail, underrun;
+ int ret = btr_node_status(wn->btrn, wn->min_iqs, BTR_NT_LEAF);
- if (!pad->handle)
- return 1;
- if (*wng->loaded - wn->written < pad->bytes_per_frame)
- return 1;
+ if (ret == 0)
+ return;
+ if (ret < 0 || !pad)
+ return sched_min_delay(s);
/*
* Data is available to be written to the alsa handle. Compute number
* of milliseconds until next buffer underrun would occur.
underrun = 50;
underrun -= 50;
ms2tv(underrun, &tv);
- if (tv_diff(&s->timeout, &tv, NULL) > 0)
- s->timeout = tv;
- return 1;
+ sched_request_timeout(&tv, s);
+}
+
+static void alsa_close(struct writer_node *wn)
+{
+ struct private_alsa_write_data *pad = wn->private_data;
+ PARA_INFO_LOG("closing writer node %p\n", wn);
+
+ if (!pad)
+ return;
+ /*
+ * It's OK to have a blocking operation here because we already made
+ * sure that the PCM output buffer is (nearly) empty.
+ */
+ snd_pcm_nonblock(pad->handle, 0);
+ snd_pcm_drain(pad->handle);
+ snd_pcm_close(pad->handle);
+ snd_config_update_free_global();
+ free(pad);
}
-static int alsa_write_post_select(__a_unused struct sched *s,
- struct writer_node *wn)
+static void alsa_write_post_select(__a_unused struct sched *s,
+ struct task *t)
{
+ struct writer_node *wn = container_of(t, struct writer_node, task);
struct private_alsa_write_data *pad = wn->private_data;
- struct writer_node_group *wng = wn->wng;
- size_t bytes = *wng->loaded - wn->written;
- unsigned char *data = (unsigned char*)*wng->bufp + wn->written;
- snd_pcm_sframes_t ret, frames, avail;
+ struct btr_node *btrn = wn->btrn;
+ char *data;
+ size_t bytes;
+ snd_pcm_sframes_t frames;
+ int ret;
- if (*wng->input_error < 0 && (!pad->handle || bytes < pad->bytes_per_frame)) {
- wn->written = *wng->loaded;
- return *wng->input_error;
+again:
+ t->error = 0;
+ ret = btr_node_status(btrn, wn->min_iqs, BTR_NT_LEAF);
+ if (ret == 0)
+ return;
+ btr_merge(btrn, wn->min_iqs);
+ bytes = btr_next_buffer(btrn, &data);
+ if (ret < 0 || bytes < wn->min_iqs) { /* eof */
+ assert(btr_no_parent(btrn));
+ ret = -E_ALSA_EOF;
+ if (!pad)
+ goto err;
+ /* wait until pending frames are played */
+ if (pad->drain_barrier.tv_sec == 0) {
+ PARA_DEBUG_LOG("waiting for device to drain\n");
+ tv_add(now, &(struct timeval)EMBRACE(0, 200 * 1000),
+ &pad->drain_barrier);
+ return;
+ }
+ if (tv_diff(now, &pad->drain_barrier, NULL) > 0)
+ goto err;
+ return;
}
- if (!bytes) /* no data available */
- return 0;
- if (!pad->handle) {
- int err = alsa_init(pad, wn->conf);
- if (err < 0)
- return err;
+ if (!pad) {
+ int32_t val;
+
+ pad = para_calloc(sizeof(*pad));
+ wn->private_data = pad;
+ if (bytes == 0) /* no data available */
+ return;
+ get_btr_sample_rate(btrn, &val);
+ pad->sample_rate = val;
+ get_btr_channels(btrn, &val);
+ pad->channels = val;
+ get_btr_sample_format(btrn, &val);
+ pad->sample_format = get_alsa_pcm_format(val);
+
+ PARA_INFO_LOG("%d channel(s), %dHz\n", pad->channels,
+ pad->sample_rate);
+ ret = alsa_init(pad, wn->conf);
+ if (ret < 0)
+ goto err;
+ wn->min_iqs = pad->bytes_per_frame;
+ goto again;
}
frames = bytes / pad->bytes_per_frame;
- if (!frames) /* less than a single frame available */
- return 0;
- avail = snd_pcm_avail_update(pad->handle);
- if (avail <= 0)
- return 0;
- frames = PARA_MIN(frames, avail);
- ret = snd_pcm_writei(pad->handle, data, frames);
- if (ret >= 0) {
- wn->written += ret * pad->bytes_per_frame;
- return 1;
+ frames = snd_pcm_writei(pad->handle, data, frames);
+ if (frames >= 0) {
+ btr_consume(btrn, frames * pad->bytes_per_frame);
+ goto again;
}
- PARA_WARNING_LOG("%s\n", snd_strerror(-ret));
- if (ret == -EPIPE) {
+ if (frames == -EPIPE) {
+ PARA_WARNING_LOG("underrun (tried to write %zu bytes)\n", bytes);
snd_pcm_prepare(pad->handle);
- return 0;
+ return;
}
- if (ret == -EAGAIN)
- return 0;
- return -E_ALSA_WRITE;
+ if (frames == -EAGAIN)
+ return;
+ PARA_WARNING_LOG("%s\n", snd_strerror(-frames));
+ ret = -E_ALSA_WRITE;
+err:
+ assert(ret < 0);
+ btr_remove_node(btrn);
+ t->error = ret;
}
-static void alsa_close(struct writer_node *wn)
+__malloc static void *alsa_parse_config_or_die(const char *options)
{
- struct private_alsa_write_data *pad = wn->private_data;
- PARA_INFO_LOG("closing writer node %p\n", wn);
+ struct alsa_write_args_info *conf = para_calloc(sizeof(*conf));
- if (pad->handle) {
- snd_pcm_drain(pad->handle);
- snd_pcm_close(pad->handle);
- snd_config_update_free_global();
- }
- free(pad);
+ PARA_INFO_LOG("options: %s, %zd\n", options, strcspn(options, " \t"));
+ /* exits on errors */
+ alsa_cmdline_parser_string(options, conf, "alsa_write");
+ return conf;
}
-__malloc static void *alsa_parse_config(const char *options)
+static void alsa_free_config(void *conf)
{
- int ret;
- struct alsa_write_args_info *conf
- = para_calloc(sizeof(struct alsa_write_args_info));
-
- PARA_INFO_LOG("options: %s, %zd\n", options, strcspn(options, " \t"));
- ret = alsa_cmdline_parser_string(options, conf, "alsa_write");
- if (ret)
- goto err_out;
- PARA_INFO_LOG("help given: %d\n", conf->help_given);
- return conf;
-err_out:
- free(conf);
- return NULL;
+ alsa_cmdline_parser_free(conf);
}
/**
- * the init function of the alsa writer
+ * The init function of the alsa writer.
*
- * \param w pointer to the writer to initialize
+ * \param w Pointer to the writer to initialize.
*
- * \sa struct writer
+ * \sa struct \ref writer.
*/
void alsa_write_init(struct writer *w)
{
struct alsa_write_args_info dummy;
alsa_cmdline_parser_init(&dummy);
- w->open = alsa_open;
w->close = alsa_close;
w->pre_select = alsa_write_pre_select;
w->post_select = alsa_write_post_select;
- w->parse_config = alsa_parse_config;
+ w->parse_config_or_die = alsa_parse_config_or_die;
w->shutdown = NULL; /* nothing to do */
+ w->free_config = alsa_free_config;
w->help = (struct ggo_help) {
.short_help = alsa_write_args_info_help,
.detailed_help = alsa_write_args_info_detailed_help
/*
- * Copyright (C) 2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2009-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file amp_filter.c Paraslash's amplify filter. */
+#include <regex.h>
+#include <stdbool.h>
+
#include "para.h"
#include "amp_filter.cmdline.h"
#include "list.h"
#include "sched.h"
#include "ggo.h"
+#include "buffer_tree.h"
#include "filter.h"
#include "string.h"
#include "error.h"
-/** The size of the output data buffer. */
-#define AMP_CHUNK_SIZE 40960
-
extern char *stat_item_values[NUM_STAT_ITEMS];
/** Data specific to the amplify filter. */
unsigned amp;
};
-static ssize_t amp_convert(char *inbuf, size_t inbuf_len, struct filter_node *fn)
-{
- size_t i, length = PARA_MIN((inbuf_len / 2) * 2,
- (fn->bufsize - fn->loaded) / 2 * 2);
- struct private_amp_data *pad = fn->private_data;
- int16_t *ip = (int16_t *)inbuf, *op = (int16_t *)(fn->buf + fn->loaded);
-
- if (!length)
- return 0;
- for (i = 0; i < length / 2; i++) {
- int x = (PARA_ABS(*ip) * (64 + pad->amp)) >> 6;
- *op++ = *ip++ > 0? PARA_MIN(x, 32767) : PARA_MAX(-x, -32768);
- }
- fn->loaded += length;
- return length;
-}
-
static void amp_close(struct filter_node *fn)
{
free(fn->private_data);
- free(fn->buf);
}
static int amp_parse_config(int argc, char **argv, void **config)
ret = -ERRNO_TO_PARA_ERROR(EINVAL);
if (amp_conf->amp_arg < 0)
goto err;
- PARA_NOTICE_LOG("amplification: %u (scaling factor: %1.2f)\n",
- amp_conf->amp_arg, amp_conf->amp_arg / 64.0 + 1.0);
*config = amp_conf;
return 1;
err:
pad->conf = fn->conf;
fn->private_data = pad;
- if (!pad->conf->amp_given && stat_item_values[SI_AMPLIFICATION]) {
- int i = SI_AMPLIFICATION;
- char *s = stat_item_values[i] + strlen(status_item_list[i]) + 1;
- sscanf(s, "%u", &pad->amp);
- } else
+ fn->min_iqs = 2;
+ if (!pad->conf->amp_given && stat_item_values[SI_AMPLIFICATION])
+ sscanf(stat_item_values[SI_AMPLIFICATION], "%u", &pad->amp);
+ else
pad->amp = pad->conf->amp_arg;
- fn->bufsize = AMP_CHUNK_SIZE;
- fn->buf = para_malloc(fn->bufsize);
+ PARA_NOTICE_LOG("amplification: %u (scaling factor: %1.2f)\n",
+ pad->amp, pad->amp / 64.0 + 1.0);
+}
+
+static void amp_post_select(__a_unused struct sched *s, struct task *t)
+{
+ struct filter_node *fn = container_of(t, struct filter_node, task);
+ struct private_amp_data *pad = fn->private_data;
+ struct btr_node *btrn = fn->btrn;
+ int ret, factor = 64 + pad->amp;
+ size_t i, in_bytes, len;
+ int16_t *in, *out;
+ bool inplace = btr_inplace_ok(btrn);
+
+ if (pad->amp == 0) { /* no amplification */
+ t->error = -E_AMP_ZERO_AMP;
+ btr_splice_out_node(btrn);
+ return;
+ }
+next_buffer:
+ ret = btr_node_status(btrn, fn->min_iqs, BTR_NT_INTERNAL);
+ if (ret < 0)
+ goto err;
+ if (ret == 0)
+ return;
+ btr_merge(btrn, fn->min_iqs);
+ in_bytes = btr_next_buffer(btrn, (char **)&in);
+ len = in_bytes / 2;
+ if (len == 0) { /* eof and in_bytes == 1 */
+ ret = -E_AMP_EOF;
+ goto err;
+ }
+
+ if (inplace)
+ out = in;
+ else
+ out = para_malloc(len * 2);
+
+ for (i = 0; i < len; i++) {
+ int x = (in[i] * factor) >> 6;
+
+ out[i] = x;
+ if (out[i] != x) /* overflow, clip */
+ out[i] = (x >= 0)? 32767 : -32768;
+ }
+
+ if (inplace)
+ btr_pushdown_one(btrn);
+ else {
+ btr_consume(btrn, len * 2);
+ btr_add_output((char *)out, len * 2, btrn);
+ }
+ t->error = 0;
+ goto next_buffer;
+err:
+ assert(ret < 0);
+ t->error = ret;
+ btr_remove_node(btrn);
+}
+
+static void amp_free_config(void *conf)
+{
+ amp_cmdline_parser_free(conf);
}
/**
amp_cmdline_parser_init(&dummy);
f->open = amp_open;
f->close = amp_close;
- f->convert = amp_convert;
+ f->pre_select = generic_filter_pre_select;
+ f->post_select = amp_post_select;
f->parse_config = amp_parse_config;
+ f->free_config = amp_free_config;
f->help = (struct ggo_help) {
.short_help = amp_filter_args_info_help,
.detailed_help = amp_filter_args_info_detailed_help
/*
- * Copyright (C) 1997-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 1997-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file attribute.c Attribute handling functions. */
+
+#include <regex.h>
+#include <openssl/rc4.h>
+#include <osl.h>
+
#include "para.h"
#include "error.h"
+#include "crypt.h"
#include "string.h"
#include "afh.h"
#include "afs.h"
struct osl_object obj = {.data = (char *)att_name,
.size = strlen(att_name) + 1};
struct osl_row *row;
- int ret = osl_get_row(attribute_table, ATTCOL_NAME, &obj, &row);
+ int ret = osl(osl_get_row(attribute_table, ATTCOL_NAME, &obj, &row));
if (ret < 0)
return ret;
- ret = osl_get_object(attribute_table, row, ATTCOL_BITNUM, &obj);
+ ret = osl(osl_get_object(attribute_table, row, ATTCOL_BITNUM, &obj));
if (ret < 0)
return ret;
*bitnum = *(unsigned char *)obj.data;
if (!(laad->flags & LSATT_FLAG_LONG))
return para_printf(&laad->pb, "%s\n", name);
- ret = osl_get_object(table, row, ATTCOL_BITNUM, &bitnum_obj);
+ ret = osl(osl_get_object(table, row, ATTCOL_BITNUM, &bitnum_obj));
if (ret < 0) {
para_printf(&laad->pb, "%s: %s\n", name, para_strerror(-ret));
return ret;
free(laad.pb.buf);
}
-int com_lsatt(int fd, int argc, char * const * const argv)
+int com_lsatt(struct rc4_context *rc4c, int argc, char * const * const argv)
{
unsigned flags = 0;
struct osl_object options = {.data = &flags, .size = sizeof(flags)};
}
}
ret = send_option_arg_callback_request(&options, argc - i, argv + i,
- com_lsatt_callback, send_result, &fd);
+ com_lsatt_callback, rc4_send_result, rc4c);
if (!ret) {
if (argc > 1)
- ret = send_va_buffer(fd, "no matches\n");
+ ret = rc4_send_va_buffer(rc4c, "no matches\n");
} else if (ret < 0)
- send_va_buffer(fd, "%s\n", para_strerror(-ret));
+ rc4_send_va_buffer(rc4c, "%s\n", para_strerror(-ret));
return ret;
}
p[len - 1] = '\0';
obj.data = p;
obj.size = len + 1;
- ret = osl_get_row(attribute_table, ATTCOL_NAME, &obj, &row);
+ ret = osl(osl_get_row(attribute_table, ATTCOL_NAME, &obj, &row));
if (ret < 0)
goto out;
- ret = osl_get_object(attribute_table, row, ATTCOL_BITNUM,
- &obj);
+ ret = osl(osl_get_object(attribute_table, row, ATTCOL_BITNUM,
+ &obj));
if (ret < 0)
goto out;
if (c == '+')
PARA_NOTICE_LOG("%s\n", para_strerror(-ret));
}
-int com_setatt(__a_unused int fd, int argc, char * const * const argv)
+int com_setatt(__a_unused struct rc4_context *rc4c, int argc, char * const * const argv)
{
if (argc < 3)
return -E_ATTR_SYNTAX;
goto out;
continue;
}
- if (ret != -E_RB_KEY_NOT_FOUND) /* error */
+ if (ret != -OSL_ERRNO_TO_PARA_ERROR(E_OSL_RB_KEY_NOT_FOUND)) /* error */
goto out;
objs[ATTCOL_BITNUM].size = 1;
/* find smallest unused attribute */
for (bitnum = 0; bitnum < 64; bitnum++) {
objs[ATTCOL_BITNUM].data = &bitnum;
- ret = osl_get_row(attribute_table, ATTCOL_BITNUM,
- &objs[ATTCOL_BITNUM], &row);
- if (ret == -E_RB_KEY_NOT_FOUND)
+ ret = osl(osl_get_row(attribute_table, ATTCOL_BITNUM,
+ &objs[ATTCOL_BITNUM], &row));
+ if (ret == -OSL_ERRNO_TO_PARA_ERROR(E_OSL_RB_KEY_NOT_FOUND))
break; /* this bitnum is unused, use it */
if (ret < 0) /* error */
goto out;
}
objs[ATTCOL_NAME].data = p;
objs[ATTCOL_NAME].size = len + 1;
- ret = osl_add_row(attribute_table, objs);
+ ret = osl(osl_add_row(attribute_table, objs));
if (ret < 0)
goto out;
aed.name = p;
free(pb.buf);
}
-int com_addatt(int fd, int argc, char * const * const argv)
+int com_addatt(struct rc4_context *rc4c, int argc, char * const * const argv)
{
int ret;
if (argc < 2)
return -E_ATTR_SYNTAX;
ret = send_standard_callback_request(argc - 1, argv + 1, com_addatt_callback,
- send_result, &fd);
+ rc4_send_result, rc4c);
if (ret < 0)
- send_va_buffer(fd, "%s\n", para_strerror(-ret));
+ rc4_send_va_buffer(rc4c, "%s\n", para_strerror(-ret));
return ret;
}
};
int ret;
- ret = osl_get_row(attribute_table, ATTCOL_NAME, &obj, &row);
+ ret = osl(osl_get_row(attribute_table, ATTCOL_NAME, &obj, &row));
if (ret < 0)
goto out;
obj.data = new;
obj.size = strlen(new) + 1;
- ret = osl_update_object(attribute_table, row, ATTCOL_NAME, &obj);
+ ret = osl(osl_update_object(attribute_table, row, ATTCOL_NAME, &obj));
out:
if (ret < 0)
para_printf(&pb, "%s\n", para_strerror(-ret));
free(pb.buf);
}
-int com_mvatt(int fd, int argc, char * const * const argv)
+int com_mvatt(struct rc4_context *rc4c, int argc, char * const * const argv)
{
int ret;
if (argc != 3)
return -E_ATTR_SYNTAX;
ret = send_standard_callback_request(argc - 1, argv + 1, com_mvatt_callback,
- send_result, &fd);
+ rc4_send_result, rc4c);
if (ret < 0)
- send_va_buffer(fd, "%s\n", para_strerror(-ret));
+ rc4_send_va_buffer(rc4c, "%s\n", para_strerror(-ret));
return ret;
}
ret = get_attribute_bitnum_by_name(name, &red.bitnum);
if (ret < 0)
return para_printf(&raad->pb, "%s: %s\n", name, para_strerror(-ret));
- ret = osl_del_row(table, row);
+ ret = osl(osl_del_row(table, row));
if (ret < 0)
return para_printf(&raad->pb, "%s: %s\n", name, para_strerror(-ret));
ret = para_printf(&raad->pb, "removed attribute %s\n", name);
free(raad.pb.buf);
}
-int com_rmatt(int fd, int argc, char * const * const argv)
+int com_rmatt(struct rc4_context *rc4c, int argc, char * const * const argv)
{
int ret;
if (argc < 2)
return -E_ATTR_SYNTAX;
ret = send_standard_callback_request(argc - 1, argv + 1, com_rmatt_callback,
- send_result, &fd);
+ rc4_send_result, rc4c);
if (ret < 0)
- send_va_buffer(fd, "%s\n", para_strerror(-ret));
+ rc4_send_va_buffer(rc4c, "%s\n", para_strerror(-ret));
return ret;
}
if (!(*atts & (one << i)))
continue;
- ret = osl_get_row(attribute_table, ATTCOL_BITNUM, &obj, &row);
+ ret = osl(osl_get_row(attribute_table, ATTCOL_BITNUM, &obj, &row));
if (ret < 0)
goto err;
- ret = osl_get_object(attribute_table, row, ATTCOL_NAME, &obj);
+ ret = osl(osl_get_object(attribute_table, row, ATTCOL_NAME, &obj));
if (ret < 0)
goto err;
if (*text) {
*
* \sa osl_close_table().
*/
-void attribute_close(void)
+static void attribute_close(void)
{
osl_close_table(attribute_table, OSL_MARK_CLEAN);
attribute_table = NULL;
int ret;
attribute_table_desc.dir = dir;
- ret = osl_open_table(&attribute_table_desc, &attribute_table);
+ ret = osl(osl_open_table(&attribute_table_desc, &attribute_table));
greatest_att_bitnum = -1; /* no atts available */
if (ret >= 0) {
find_greatest_att_bitnum();
return ret;
}
attribute_table = NULL;
- if (ret >= 0 || is_errno(-ret, ENOENT))
+ if (ret >= 0 || ret == -OSL_ERRNO_TO_PARA_ERROR(E_OSL_NOENT))
return 1;
return ret;
}
static int attribute_create(const char *dir)
{
attribute_table_desc.dir = dir;
- return osl_create_table(&attribute_table_desc);
+ return osl(osl_create_table(&attribute_table_desc));
}
/**
/*
- * Copyright (C) 2005-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2005-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file audioc.c the client program used to connect to para_audiod */
+#include <regex.h>
#include <sys/types.h>
#include <dirent.h>
INIT_AUDIOC_ERRLISTS;
-/** the gengetopt structure containing command line args */
-struct audioc_args_info conf;
+/** The gengetopt structure containing command line args. */
+static struct audioc_args_info conf;
static int loglevel;
INIT_STDERR_LOGGING(loglevel);
{
int ret = -E_AUDIOC_SYNTAX, fd;
char *cf, *buf = NULL, *args;
- size_t bufsize, loaded = 0;
+ size_t bufsize;
if (audioc_cmdline_parser(argc, argv, &conf))
goto out;
args = conf.inputs_num?
concat_args(conf.inputs_num, conf.inputs) :
para_strdup("stat");
- bufsize = conf.bufsize_arg;
- buf = para_malloc(bufsize);
- if (conf.socket_given) {
- ret = create_remote_socket(conf.socket_arg);
- } else {
- char *hn = para_hostname(),
- *socket_name = make_message("/var/paraslash/audiod_socket.%s", hn);
-
- ret = create_remote_socket(socket_name);
+ if (conf.socket_given)
+ ret = connect_local_socket(conf.socket_arg);
+ else {
+ char *hn = para_hostname(), *socket_name = make_message(
+ "/var/paraslash/audiod_socket.%s", hn);
+ ret = connect_local_socket(socket_name);
free(hn);
free(socket_name);
}
- if (ret < 0)
+ if (ret < 0) {
+ PARA_EMERG_LOG("failed to connect to local socket\n");
goto out;
+ }
fd = ret;
- ret = mark_fd_nonblocking(fd);
- if (ret < 0)
- goto out;
- ret = mark_fd_nonblocking(STDOUT_FILENO);
- if (ret < 0)
- goto out;
ret = send_cred_buffer(fd, args);
if (ret < 0)
goto out;
- for (;;) {
- int max_fileno = -1, check_write = 0;
- ssize_t len;
- fd_set rfd, wfd;
- FD_ZERO(&rfd);
- FD_ZERO(&wfd);
- if (loaded < bufsize)
- para_fd_set(fd, &rfd, &max_fileno);
- if (loaded > 0) {
- para_fd_set(STDOUT_FILENO, &wfd, &max_fileno);
- check_write = 1;
- }
- ret = -E_AUDIOC_OVERRUN;
- if (max_fileno < 0)
- goto out;
- ret = para_select(max_fileno + 1, &rfd, &wfd, NULL);
- if (ret < 0)
- goto out;
- if (loaded < bufsize && FD_ISSET(fd, &rfd)) {
- len = recv_bin_buffer(fd, buf + loaded,
- bufsize - loaded);
- if (len <= 0) {
- ret = len < 0? -E_AUDIOC_READ : 0;
- goto out;
- }
- loaded += len;
- }
- if (check_write && FD_ISSET(STDOUT_FILENO, &wfd)) {
- ret = write(STDOUT_FILENO, buf, loaded);
- if (ret < 0) {
- ret = -E_AUDIOC_WRITE;
- goto out;
- }
- loaded -= ret;
- if (loaded && ret)
- memmove(buf, buf + ret, loaded);
- }
- }
+ bufsize = conf.bufsize_arg;
+ buf = para_malloc(bufsize);
+ do {
+ size_t n = ret = recv_bin_buffer(fd, buf, bufsize);
+ if (ret <= 0)
+ break;
+ ret = write_all(STDOUT_FILENO, buf, &n);
+ } while (ret >= 0);
out:
- if (!ret && loaded && buf)
- ret = write(STDOUT_FILENO, buf, loaded);
if (ret < 0)
PARA_ERROR_LOG("%s\n", para_strerror(-ret));
return ret < 0? EXIT_FAILURE : EXIT_SUCCESS;
/*
- * Copyright (C) 2005-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2005-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
/** \file audiod.c the paraslash's audio daemon */
+#include <regex.h>
#include <sys/types.h>
#include <dirent.h>
#include <signal.h>
+#include <openssl/rc4.h>
+#include <stdbool.h>
#include "para.h"
#include "error.h"
+#include "crypt.h"
#include "audiod.cmdline.h"
#include "list.h"
#include "sched.h"
#include "ggo.h"
#include "recv.h"
+#include "buffer_tree.h"
#include "filter.h"
-#include "grab_client.cmdline.h"
#include "grab_client.h"
#include "client.cmdline.h"
#include "client.h"
struct timeval restart_barrier;
/** Last time we received status data from para_server. */
struct timeval last_status_read;
+ size_t min_iqs;
/** The offset value announced by para_server. */
int offset_seconds;
/** The length of the current audio file as announced by para_server. */
struct timeval clock_diff_barrier;
/** Number of the audio format as announced by para_server. */
int current_audio_format_num;
+ /* The status task btrn is the child of the client task. */
+ struct btr_node *btrn;
};
/** The array of status items sent by para_server. */
* \sa struct status_task
*/
static struct status_task *stat_task = &status_task_struct;
-static struct timeval initial_delay_barrier;
/**
* the task for handling audiod commands
#define FOR_EACH_AUDIO_FORMAT(af) for (af = 0; af < NUM_AUDIO_FORMATS; af++)
/**
- * get the audio format number
- * \param name the name of the audio format
+ * Get the audio format number.
+ *
+ * \param name The name of the audio format.
*
* \return The audio format number on success, -E_UNSUPPORTED_AUDIO_FORMAT if
* \a name is not a supported audio format.
*/
-int get_audio_format_num(char *name)
+int get_audio_format_num(const char *name)
{
int i;
return -E_UNSUPPORTED_AUDIO_FORMAT;
}
+/**
+ * Compute the play time based on information of the given slot.
+ *
+ * \param slot_num The slot number (negative means: no slot).
+ *
+ * This computes a string of the form "0:07 [3:33] (3%/3:40)" using information
+ * from the status items received from para_server and the start time of the
+ * (first) writer of the given slot.
+ *
+ * It has to to take into account that probably the stream was not started at
+ * the beginning of the file, that the clock between the server and the client
+ * host may differ and that playback of the stream was delayed, e.g. because
+ * the prebuffer filter is used in the filter configuration of the given slot.
+ *
+ * If no writer is active in the given slot, or \a slot_num is negative
+ * (indicating that para_audiod runs in standby mode), an approximation based
+ * only on the status items is computed and the returned string is prefixed
+ * with "~".
+ *
+ * \return A string that must be freed by the caller.
+ */
char *get_time_string(int slot_num)
{
int ret, seconds = 0, length;
struct timeval *tmp, sum, sss, /* server stream start */
+ rstime, /* receiver start time */
+ wstime, /* writer start time */
wtime, /* now - writer start */
rskip; /* receiver start - sss */
struct slot_info *s = slot_num < 0? NULL : &slot[slot_num];
+ char *msg;
if (audiod_status == AUDIOD_OFF)
goto empty;
}
if (audiod_status == AUDIOD_ON && !s)
goto empty;
- /* valid status items and playing */
- if (s) { /* writer active in this slot */
- length = s->seconds_total;
- tmp = &s->server_stream_start;
- } else { /* standby mode, rely on status items */
- length = stat_task->length_seconds;
- tmp = &stat_task->server_stream_start;
+ /*
+ * Valid status items and playing, set length and tmp to the stream
+ * start. We use the slot info and fall back to the info from current
+ * status items if no slot info is available.
+ */
+ length = stat_task->length_seconds;
+ tmp = &stat_task->server_stream_start;
+ if (s && s->wns) { /* writer active in this slot */
+ btr_get_node_start(s->wns[0].btrn, &wstime);
+ if (wstime.tv_sec != 0) { /* writer wrote something */
+ if (s->server_stream_start.tv_sec == 0) {
+ /* copy status info to slot */
+ s->server_stream_start = stat_task->server_stream_start;
+ s->offset_seconds = stat_task->offset_seconds;
+ s->seconds_total = stat_task->length_seconds;
+ }
+ length = s->seconds_total;
+ tmp = &s->server_stream_start;
+ }
}
if (stat_task->sa_time_diff_sign > 0)
tv_diff(tmp, &stat_task->sa_time_diff, &sss);
else
tv_add(tmp, &stat_task->sa_time_diff, &sss);
- if (!s) {
+ if (!s || !s->wns) {
struct timeval diff;
tv_diff(now, &sss, &diff);
seconds = diff.tv_sec + stat_task->offset_seconds;
goto out;
}
- tv_diff(now, &s->wstime, &wtime);
+ tv_diff(now, &wstime, &wtime);
+ //PARA_CRIT_LOG("offset %d\n", s->offset_seconds);
seconds = s->offset_seconds;
- ret = tv_diff(&s->rstime, &sss, &rskip);
+ btr_get_node_start(s->receiver_node->btrn, &rstime);
+ ret = tv_diff(&rstime, &sss, &rskip);
if (ret > 0) { /* audiod was started in the middle of the stream */
tv_add(&wtime, &rskip, &sum);
seconds += sum.tv_sec;
out:
seconds = PARA_MIN(seconds, length);
seconds = PARA_MAX(seconds, 0);
- return make_message(
- "%s: %s%d:%02d [%d:%02d] (%d%%/%d:%02d)\n",
- status_item_list[SI_PLAY_TIME],
+ msg = make_message(
+ "%s%d:%02d [%d:%02d] (%d%%/%d:%02d)",
s? "" : "~",
seconds / 60,
seconds % 60,
length / 60,
length % 60
);
+ PARA_DEBUG_LOG("slot %d: %s\n", slot_num, msg);
+ return msg;
empty:
- return make_message("%s:\n", status_item_list[SI_PLAY_TIME]);
+ return para_strdup(NULL);
}
static int want_colors(void)
PARA_NOTICE_LOG("closing %s receiver in slot %d\n",
audio_formats[s->format], slot_num);
a->receiver->close(s->receiver_node);
+ btr_free_node(s->receiver_node->btrn);
free(s->receiver_node);
s->receiver_node = NULL;
- stat_task->current_audio_format_num = -1;
+ tv_add(now, &(struct timeval)EMBRACE(0, 200 * 1000),
+ &a->restart_barrier);
}
-static void kill_all_decoders(int error)
+static void writer_cleanup(struct writer_node *wn)
+{
+ struct writer *w;
+
+ if (!wn)
+ return;
+ w = writers + wn->writer_num;
+ PARA_INFO_LOG("closing %s\n", writer_names[wn->writer_num]);
+ w->close(wn);
+ btr_free_node(wn->btrn);
+}
+
+static void close_writers(struct slot_info *s)
+{
+ struct audio_format_info *a;
+ int i;
+
+ if (s->format < 0)
+ return;
+ assert(s->wns);
+ a = afi + s->format;
+ if (a->num_writers == 0)
+ writer_cleanup(s->wns);
+ else {
+ for (i = 0; i < a->num_writers; i++)
+ writer_cleanup(s->wns + i);
+ }
+ free(s->wns);
+ s->wns = NULL;
+}
+
+static void close_filters(struct slot_info *s)
{
int i;
+ struct audio_format_info *a = afi + s->format;
+ if (a->num_filters == 0)
+ return;
+ for (i = 0; i < a->num_filters; i++) {
+ struct filter_node *fn = s->fns + i;
+ struct filter *f;
+
+ if (!fn)
+ continue;
+ f = filters + fn->filter_num;
+ if (f->close)
+ f->close(fn);
+ btr_free_node(fn->btrn);
+ }
+ free(s->fns);
+ s->fns = NULL;
+}
+
+/*
+ * Whenever a task commits suicide by returning from post_select with t->error
+ * < 0, it also removes its btr node. We do exactly that to kill a running
+ * task. Note that the scheduler checks t->error also _before_ each pre/post
+ * select call, so the victim will never be scheduled again.
+ */
+static void kill_btrn(struct btr_node *btrn, struct task *t, int error)
+{
+ if (t->error < 0)
+ return;
+ t->error = error;
+ btr_remove_node(btrn);
+}
+
+static void kill_all_decoders(int error)
+{
+ int i, j;
FOR_EACH_SLOT(i) {
struct slot_info *s = &slot[i];
- if (s->wng && s->wng->task.error >= 0) {
- PARA_INFO_LOG("deactivating wng in slot %d\n",
- i);
- s->wng->task.error = error;
- }
- if (s->fc && s->fc->task.error >= 0) {
- PARA_INFO_LOG("deactivatimg filter chain in slot %d\n", i);
- s->fc->task.error = error;
- }
- if (s->receiver_node && s->receiver_node->task.error >= 0) {
- PARA_INFO_LOG("deactivating receiver_node in slot %d\n", i);
- s->receiver_node->task.error = error;
- }
+ struct audio_format_info *a;
+ if (s->format < 0)
+ continue;
+ a = afi + s->format;
+ if (s->wns)
+ for (j = 0; j < a->num_writers; j++)
+ kill_btrn(s->wns[j].btrn, &s->wns[j].task, error);
+ if (s->fns)
+ for (j = 0; j < a->num_writers; j++)
+ kill_btrn(s->fns[j].btrn, &s->wns[j].task, error);
+ if (s->receiver_node)
+ kill_btrn(s->receiver_node->btrn, &s->receiver_node->task,
+ error);
}
}
clear_slot(i);
return i;
}
- if (s->wng || s->receiver_node || s->fc)
+ if (s->wns || s->receiver_node || s->fns)
continue;
clear_slot(i);
return i;
return afi[audio_format_num].num_filters;
}
-static void open_filters(int slot_num)
+static void open_filters(struct slot_info *s)
{
- struct slot_info *s = &slot[slot_num];
- struct audio_format_info *a = &afi[s->format];
+ struct audio_format_info *a = afi + s->format;
struct filter_node *fn;
int nf = a->num_filters;
+ struct btr_node *parent;
int i;
- s->fc = NULL;
- if (!nf)
+ if (nf == 0)
return;
PARA_INFO_LOG("opening %s filters\n", audio_formats[s->format]);
- s->fc = para_calloc(sizeof(struct filter_chain));
- s->fc->filter_nodes = para_malloc(nf * sizeof(struct filter_node));
- s->fc->inbufp = &s->receiver_node->buf;
- s->fc->in_loaded = &s->receiver_node->loaded;
- s->fc->input_error = &s->receiver_node->task.error;
- s->fc->task.pre_select = NULL;
- s->fc->task.post_select = filter_post_select;
- s->fc->task.error = 0;
- s->fc->num_filters = nf;
-
- s->receiver_node->output_error = &s->fc->task.error;
- sprintf(s->fc->task.status, "filter chain");
- FOR_EACH_FILTER_NODE(fn, s->fc, i) {
+ assert(s->fns == NULL);
+ s->fns = para_calloc(nf * sizeof(struct filter_node));
+ parent = s->receiver_node->btrn;
+ for (i = 0; i < nf; i++) {
struct filter *f = filters + a->filter_nums[i];
+ fn = s->fns + i;
fn->filter_num = a->filter_nums[i];
fn->conf = a->filter_conf[i];
- fn->fc = s->fc;
- fn->loaded = 0;
- INIT_LIST_HEAD(&fn->callbacks);
+ fn->task.pre_select = f->pre_select;
+ fn->task.post_select = f->post_select;
+
+ fn->btrn = btr_new_node(&(struct btr_node_description)
+ EMBRACE(.name = f->name, .parent = parent,
+ .handler = f->execute, .context = fn));
+
f->open(fn);
+ register_task(&fn->task);
+ parent = fn->btrn;
PARA_NOTICE_LOG("%s filter %d/%d (%s) started in slot %d\n",
- audio_formats[s->format], i, nf, f->name, slot_num);
- s->fc->outbufp = &fn->buf;
- s->fc->out_loaded = &fn->loaded;
+ audio_formats[s->format], i, nf, f->name, (int)(s - slot));
+ sprintf(fn->task.status, "%s (slot %d)", f->name, (int)(s - slot));
}
- register_task(&s->fc->task);
}
-static void open_writers(int slot_num)
+static void open_writers(struct slot_info *s)
{
- int ret, i;
- struct slot_info *s = &slot[slot_num];
- struct audio_format_info *a = &afi[s->format];
-
- PARA_INFO_LOG("opening %s writers\n", audio_formats[s->format]);
- if (!a->num_writers)
- s->wng = setup_default_wng();
- else
- s->wng = wng_new(a->num_writers);
- if (s->fc) {
- s->wng->bufp = s->fc->outbufp;
- s->wng->loaded = s->fc->out_loaded;
- s->wng->input_error = &s->fc->task.error;
- s->wng->channels = &s->fc->channels;
- s->wng->samplerate = &s->fc->samplerate;
- s->fc->output_error = &s->wng->task.error;
- PARA_INFO_LOG("samplerate: %d\n", *s->wng->samplerate);
- } else {
- s->wng->bufp = &s->receiver_node->buf;
- s->wng->loaded = &s->receiver_node->loaded;
- s->wng->input_error = &s->receiver_node->task.error;
- }
- for (i = 0; i < a->num_writers; i++) {
- s->wng->writer_nodes[i].conf = a->writer_conf[i];
- s->wng->writer_nodes[i].writer_num = a->writer_nums[i];
+ int i;
+ struct audio_format_info *a = afi + s->format;
+ struct writer_node *wn;
+ struct btr_node *parent = s->fns[a->num_filters - 1].btrn;
+
+ assert(s->wns == NULL);
+ s->wns = para_calloc(PARA_MAX(1U, a->num_writers)
+ * sizeof(struct writer_node));
+ if (a->num_writers == 0)
+ setup_writer_node(NULL, parent, s->wns);
+ else {
+ PARA_INFO_LOG("opening %s writers\n", audio_formats[s->format]);
+ for (i = 0; i < a->num_writers; i++) {
+ wn = s->wns + i;
+ wn->conf = a->writer_conf[i];
+ wn->writer_num = a->writer_nums[i];
+ register_writer_node(wn, parent);
+ }
}
- ret = wng_open(s->wng);
- if (ret < 0)
- return;
- s->wstime = *now;
- s->server_stream_start = stat_task->server_stream_start.tv_sec?
- stat_task->server_stream_start : *now;
- s->offset_seconds = stat_task->offset_seconds;
- s->seconds_total = stat_task->length_seconds;
- activate_inactive_grab_clients(slot_num, s->format, s->fc);
}
+/* returns slot num on success */
static int open_receiver(int format)
{
struct audio_format_info *a = &afi[format];
struct slot_info *s;
int ret, slot_num;
+ struct receiver *r = a->receiver;
struct receiver_node *rn;
- const struct timeval restart_delay = {2, 0};
+ tv_add(now, &(struct timeval)EMBRACE(2, 0), &a->restart_barrier);
ret = get_empty_slot();
if (ret < 0)
- goto err;
+ return ret;
slot_num = ret;
- s = &slot[slot_num];
- s->format = format;
- s->receiver_node = para_calloc(sizeof(struct receiver_node));
- rn = s->receiver_node;
- rn->receiver = a->receiver;
+ rn = para_calloc(sizeof(*rn));
+ rn->receiver = r;
rn->conf = a->receiver_conf;
- ret = a->receiver->open(s->receiver_node);
+ rn->btrn = btr_new_node(&(struct btr_node_description)
+ EMBRACE(.name = r->name, .context = rn));
+ ret = r->open(rn);
if (ret < 0) {
- free(s->receiver_node);
- s->receiver_node = NULL;
- goto err;
+ btr_free_node(rn->btrn);
+ free(rn);
+ return ret;
}
+ s = &slot[slot_num];
+ s->format = format;
+ s->receiver_node = rn;
PARA_NOTICE_LOG("started %s: %s receiver in slot %d\n",
- audio_formats[s->format], a->receiver->name, slot_num);
- rn->task.pre_select = a->receiver->pre_select;
- rn->task.post_select = a->receiver->post_select;
- s->rstime = *now;
- sprintf(rn->task.status, "%s receiver node", rn->receiver->name);
+ audio_formats[format], r->name, slot_num);
+ rn->task.pre_select = r->pre_select;
+ rn->task.post_select = r->post_select;
+ sprintf(rn->task.status, "%s receiver node", r->name);
register_task(&rn->task);
- ret = 1;
-err:
- if (ret < 0)
- PARA_ERROR_LOG("%s\n", para_strerror(-ret));
- tv_add(now, &restart_delay, &afi[format].restart_barrier);
- return ret;
+ return slot_num;
}
-/* return: 0: Not running, 1: Running, -1: Running but eof (or error) */
-static int receiver_running(int format)
+static bool receiver_running(void)
{
- int i, ret = 0;
+ int i;
+ long unsigned ss1 = stat_task->server_stream_start.tv_sec;
FOR_EACH_SLOT(i) {
struct slot_info *s = &slot[i];
- if (s->format != format)
- continue;
+ long unsigned ss2 = s->server_stream_start.tv_sec;
+
if (!s->receiver_node)
continue;
if (s->receiver_node->task.error >= 0)
- return 1;
- ret = -1;
+ return true;
+ if (ss1 == ss2)
+ return true;
}
- return ret;
+ return false;
}
-static void open_current_receiver(struct sched *s)
+/**
+ * Return the root node of the current buffer tree.
+ *
+ * This is only used for stream grabbing.
+ *
+ * \return \p NULL if no slot is currently active. If more than one buffer tree
+ * exists, the node corresponding to the most recently started receiver is
+ * returned.
+ */
+struct btr_node *audiod_get_btr_root(void)
{
- struct timeval diff;
- int ret, cafn = stat_task->current_audio_format_num;
+ int i, newest_slot = -1;
+ struct timeval newest_rstime = {0, 0};
- if (cafn < 0 || !stat_task->ct)
- return;
- /* Do nothing if the 'N' flag is set or the 'P' flag is unset */
- if (stat_task->vss_status != VSS_STATUS_FLAG_PLAYING)
- return;
- ret = receiver_running(cafn);
- if (ret > 0) /* already running and not eof */
- return;
- if (ret < 0) { /* eof */
- /*
- * para_server uses a zero start time during the announcement
- * period, i.e. before it sends the first chunk. Wait until
- * this period begins to avoid restarting the receiver that
- * belongs to the file just completed.
- */
- if (stat_task->server_stream_start.tv_sec)
- return;
- }
- if (tv_diff(now, &afi[cafn].restart_barrier, &diff) < 0) {
- /* avoid busy loop */
- s->timeout = diff;
- return;
+ FOR_EACH_SLOT(i) {
+ struct slot_info *s = &slot[i];
+ struct timeval rstime;
+ if (!s->receiver_node)
+ continue;
+ if (s->receiver_node->task.error < 0)
+ continue;
+ btr_get_node_start(s->receiver_node->btrn, &rstime);
+ if (newest_slot >= 0 && tv_diff(&rstime, &newest_rstime, NULL) < 0)
+ continue;
+ newest_rstime = rstime;
+ newest_slot = i;
}
- /* start a new receiver */
- open_receiver(cafn);
+ if (newest_slot == -1)
+ return NULL;
+ return slot[newest_slot].receiver_node->btrn;
+}
+
+/* whether a new instance of a decoder should be started. */
+static bool must_start_decoder(void)
+{
+ int cafn = stat_task->current_audio_format_num;
+ unsigned vs = stat_task->vss_status;
+
+ if (audiod_status != AUDIOD_ON)
+ return false;
+ if (cafn < 0)
+ return false;
+ if (!stat_task->ct)
+ return false;
+ if (vs & VSS_STATUS_FLAG_NEXT)
+ return false;
+ if (!(vs & VSS_STATUS_FLAG_PLAYING))
+ return false;
+ if (receiver_running())
+ return false;
+ if (tv_diff(now, &afi[cafn].restart_barrier, NULL) < 0)
+ return false;
+ return true;
}
static unsigned compute_time_diff(const struct timeval *status_time)
&tmp);
stat_task->sa_time_diff = tmp;
PARA_INFO_LOG("time diff (cur/avg): %s%lums/%s%lums\n",
- sign > 0? "+" : "-",
+ sign < 0? "-" : "+",
tv2ms(&diff),
- sa_time_diff_sign ? "+" : "-",
+ sa_time_diff_sign < 0? "-" : "+",
tv2ms(&stat_task->sa_time_diff)
);
out:
return count;
}
-static int check_stat_line(char *line, __a_unused void *data)
+static int update_item(int itemnum, char *buf)
{
- int itemnum;
- size_t ilen = 0;
long unsigned sec, usec;
- char *tmp;
- //PARA_INFO_LOG("line: %s\n", line);
- if (!line)
- return 1;
- itemnum = stat_line_valid(line);
- if (itemnum < 0) {
- PARA_WARNING_LOG("invalid status line: %s\n", line);
- return 1;
- }
if (stat_task->clock_diff_count && itemnum != SI_CURRENT_TIME)
return 1;
- tmp = make_message("%s\n", line);
- stat_client_write(tmp, itemnum);
- free(tmp);
free(stat_item_values[itemnum]);
- stat_item_values[itemnum] = para_strdup(line);
- ilen = strlen(status_item_list[itemnum]);
+ stat_item_values[itemnum] = para_strdup(buf);
+ stat_client_write_item(itemnum);
switch (itemnum) {
case SI_STATUS_FLAGS:
stat_task->vss_status = 0;
- if (strchr(line, 'N'))
+ if (strchr(buf, 'N'))
stat_task->vss_status |= VSS_STATUS_FLAG_NEXT;
- if (strchr(line, 'P'))
+ if (strchr(buf, 'P'))
stat_task->vss_status |= VSS_STATUS_FLAG_PLAYING;
break;
case SI_OFFSET:
- stat_task->offset_seconds = atoi(line + ilen + 1);
+ stat_task->offset_seconds = atoi(buf);
break;
case SI_SECONDS_TOTAL:
- stat_task->length_seconds = atoi(line + ilen + 1);
+ stat_task->length_seconds = atoi(buf);
break;
case SI_STREAM_START:
- if (sscanf(line + ilen + 1, "%lu.%lu", &sec, &usec) == 2) {
- struct timeval a_start, delay;
- delay.tv_sec = conf.stream_delay_arg / 1000;
- delay.tv_usec = (conf.stream_delay_arg % 1000) * 1000;
+ if (sscanf(buf, "%lu.%lu", &sec, &usec) == 2) {
stat_task->server_stream_start.tv_sec = sec;
stat_task->server_stream_start.tv_usec = usec;
- if (compute_time_diff(NULL) > 2) {
- if (stat_task->sa_time_diff_sign < 0)
- tv_add(&stat_task->server_stream_start,
- &stat_task->sa_time_diff, &a_start);
- else
- tv_diff(&stat_task->server_stream_start,
- &stat_task->sa_time_diff, &a_start);
- tv_add(&a_start, &delay, &initial_delay_barrier);
- }
}
break;
case SI_CURRENT_TIME:
- if (sscanf(line + ilen + 1, "%lu.%lu", &sec, &usec) == 2) {
+ if (sscanf(buf, "%lu.%lu", &sec, &usec) == 2) {
struct timeval tv = {sec, usec};
compute_time_diff(&tv);
}
break;
case SI_FORMAT:
- stat_task->current_audio_format_num = get_audio_format_num(
- line + ilen + 1);
+ stat_task->current_audio_format_num
+ = get_audio_format_num(buf);
}
return 1;
}
if (a->num_filters)
continue; /* no default -- nothing to to */
+ /*
+ * If udp is used to receive this audiod format, add fecdec as
+ * the first filter.
+ */
+ if (strcmp(afi[i].receiver->name, "udp") == 0 ||
+ strcmp(afi[i].receiver->name, "dccp") == 0) {
+ tmp = para_strdup("fecdec");
+ add_filter(i, tmp);
+ free(tmp);
+ if (ret < 0)
+ goto out;
+ }
/* add "dec" to audio format name */
tmp = make_message("%sdec", audio_formats[i]);
for (j = 0; filters[j].name; j++)
{
int i, ret, nf;
- nf = PARA_MAX(1U, conf.filter_given);
+ nf = PARA_MAX(2U, conf.filter_given);
PARA_INFO_LOG("maximal number of filters: %d\n", nf);
FOR_EACH_AUDIO_FORMAT(i) {
afi[i].filter_conf = para_malloc(nf * sizeof(void *));
para_fd_set(st->fd, &s->rfds, &s->max_fileno);
}
-static void signal_post_select(struct sched *s, struct task *t)
+static void signal_post_select(struct sched *s, __a_unused struct task *t)
{
- struct signal_task *st = container_of(t, struct signal_task, task);
+ int signum;
- if (!FD_ISSET(st->fd, &s->rfds))
- return;
-
- st->signum = para_next_signal();
- switch (st->signum) {
+ signum = para_next_signal(&s->rfds);
+ switch (signum) {
case SIGINT:
case SIGTERM:
case SIGHUP:
- PARA_EMERG_LOG("terminating on signal %d\n", st->signum);
+ PARA_EMERG_LOG("terminating on signal %d\n", signum);
clean_exit(EXIT_FAILURE, "caught deadly signal");
}
}
{
int ret;
struct command_task *ct = container_of(t, struct command_task, task);
+ static struct timeval last_status_dump;
+ struct timeval tmp, delay = {0, 500 * 1000};
- audiod_status_dump();
- if (!FD_ISSET(ct->fd, &s->rfds))
- return;
- ret = handle_connect(ct->fd);
+ tv_add(&last_status_dump, &delay, &tmp);
+ if (tv_diff(&tmp, now, NULL) < 0) {
+ audiod_status_dump();
+ last_status_dump = *now;
+ }
+
+ ret = handle_connect(ct->fd, &s->rfds);
if (ret < 0)
PARA_ERROR_LOG("%s\n", para_strerror(-ret));
+ audiod_status_dump();
}
static void init_command_task(struct command_task *ct)
static void close_stat_pipe(void)
{
- int i;
-
if (!stat_task->ct)
return;
+ btr_free_node(stat_task->ct->btrn);
client_close(stat_task->ct);
stat_task->ct = NULL;
- FOR_EACH_STATUS_ITEM(i) {
- free(stat_item_values[i]);
- stat_item_values[i] = NULL;
- }
- dump_empty_status();
+ clear_and_dump_items();
stat_task->length_seconds = 0;
stat_task->offset_seconds = 0;
stat_task->vss_status = 0;
static void try_to_close_slot(int slot_num)
{
struct slot_info *s = &slot[slot_num];
+ struct audio_format_info *a = afi + s->format;
+ int i;
if (s->format < 0)
return;
if (s->receiver_node && s->receiver_node->task.error != -E_TASK_UNREGISTERED)
return;
- if (s->fc && s->fc->task.error != -E_TASK_UNREGISTERED)
- return;
- if (s->wng && s->wng->task.error != -E_TASK_UNREGISTERED)
- return;
+ for (i = 0; i < a->num_filters; i++)
+ if (s->fns && s->fns[i].task.error != -E_TASK_UNREGISTERED)
+ return;
+ if (a->num_writers > 0) {
+ for (i = 0; i < a->num_writers; i++)
+ if (s->wns && s->wns[i].task.error != -E_TASK_UNREGISTERED)
+ return;
+ } else {
+ if (s->wns && s->wns[0].task.error != -E_TASK_UNREGISTERED)
+ return;
+ }
PARA_INFO_LOG("closing slot %d\n", slot_num);
- wng_close(s->wng);
- close_filters(s->fc);
- free(s->fc);
+ close_writers(s);
+ close_filters(s);
close_receiver(slot_num);
clear_slot(slot_num);
}
* Check if any receivers/filters/writers need to be started and do so if
* necessary.
*/
-static void start_stop_decoders(struct sched *s)
+static void start_stop_decoders(void)
{
- int i;
+ int i, ret;
+ struct slot_info *sl;
+ struct audio_format_info *a;
FOR_EACH_SLOT(i)
try_to_close_slot(i);
if (audiod_status != AUDIOD_ON ||
!(stat_task->vss_status & VSS_STATUS_FLAG_PLAYING))
return kill_all_decoders(-E_NOT_PLAYING);
- open_current_receiver(s);
- FOR_EACH_SLOT(i) {
- struct slot_info *sl = &slot[i];
- struct audio_format_info *a;
- struct timeval diff;
-
- if (sl->format < 0)
- continue;
- a = &afi[sl->format];
- if (!sl->receiver_node)
- continue;
- if ((!a->num_filters || sl->fc) && sl->wng)
- continue; /* everything already started */
- if (!a->num_filters) {
- if (sl->receiver_node->loaded && !sl->wng) {
- open_writers(i);
- }
- continue;
- }
- if (sl->receiver_node->loaded && !sl->fc) {
- open_filters(i);
- continue;
- }
- if (sl->wng || !sl->fc || !*sl->fc->out_loaded)
- continue;
- if (tv_diff(now, &initial_delay_barrier, &diff) > 0) {
- open_writers(i);
- continue;
- }
- PARA_INFO_LOG("initial delay: %lu ms left\n", tv2ms(&diff));
- if (tv_diff(&s->timeout, &diff, NULL) > 0) {
- s->timeout = diff;
- }
+ if (!must_start_decoder())
+ return;
+ ret = open_receiver(stat_task->current_audio_format_num);
+ if (ret < 0) {
+ PARA_ERROR_LOG("%s\n", para_strerror(-ret));
+ return;
}
+ sl = slot + ret;
+ a = afi + sl->format;
+ if (a->num_filters)
+ open_filters(sl);
+ open_writers(sl);
+ activate_grab_clients();
+ btr_log_tree(sl->receiver_node->btrn, LL_NOTICE);
}
+static void status_pre_select(struct sched *s, struct task *t)
+{
+ struct status_task *st = container_of(t, struct status_task, task);
+ int ret, cafn = stat_task->current_audio_format_num;
+
+ if (must_start_decoder())
+ goto min_delay;
+ ret = btr_node_status(st->btrn, 0, BTR_NT_LEAF);
+ if (ret > 0)
+ goto min_delay;
+ if (st->ct && audiod_status == AUDIOD_OFF)
+ goto min_delay;
+ if (!st->ct && audiod_status != AUDIOD_OFF)
+ sched_request_barrier_or_min_delay(&st->restart_barrier, s);
+ if (cafn >= 0)
+ sched_request_barrier(&afi[cafn].restart_barrier, s);
+ /*
+ * If para_server is playing we'd like to have a smooth time display
+ * even if we are running in standby mode. So we request a timeout that
+ * expires at the next full second.
+ */
+ if (stat_task->vss_status & VSS_STATUS_FLAG_PLAYING)
+ sched_request_timeout_ms(1000 - now->tv_usec / 1000, s);
+ return;
+min_delay:
+ sched_min_delay(s);
+}
/* restart the client task if necessary */
-static void status_pre_select(struct sched *s, struct task *t)
+static void status_post_select(__a_unused struct sched *s, struct task *t)
{
struct status_task *st = container_of(t, struct status_task, task);
if (!st->ct)
goto out;
if (st->ct->task.error >= 0) {
- st->ct->task.error = -E_AUDIOD_OFF;
+ kill_btrn(st->ct->btrn, &st->ct->task, -E_AUDIOD_OFF);
goto out;
}
if (st->ct->task.error != -E_TASK_UNREGISTERED)
goto out;
}
if (st->ct) {
- unsigned bytes_left;
+ char *buf;
+ size_t sz;
+ int ret;
if (st->ct->task.error < 0) {
if (st->ct->task.error != -E_TASK_UNREGISTERED)
goto out;
}
if (st->ct->status != CL_RECEIVING)
goto out;
- bytes_left = for_each_line(st->ct->buf, st->ct->loaded,
- &check_stat_line, NULL);
- if (st->ct->loaded != bytes_left) {
- st->last_status_read = *now;
- st->ct->loaded = bytes_left;
- } else {
+ ret = btr_node_status(st->btrn, st->min_iqs, BTR_NT_LEAF);
+ if (ret <= 0) {
struct timeval diff;
tv_diff(now, &st->last_status_read, &diff);
if (diff.tv_sec > 61)
- st->ct->task.error = -E_STATUS_TIMEOUT;
+ kill_btrn(st->ct->btrn, &st->ct->task,
+ -E_STATUS_TIMEOUT);
+ goto out;
}
+ btr_merge(st->btrn, st->min_iqs);
+ sz = btr_next_buffer(st->btrn, &buf);
+ ret = for_each_stat_item(buf, sz, update_item);
+ if (ret < 0) {
+ kill_btrn(st->ct->btrn, &st->ct->task, ret);
+ goto out;
+ }
+ if (sz != ret) {
+ btr_consume(st->btrn, sz - ret);
+ st->last_status_read = *now;
+ st->min_iqs = 0;
+ } else /* current status item crosses buffers */
+ st->min_iqs = sz + 1;
goto out;
}
+ btr_drain(st->btrn);
+ st->current_audio_format_num = -1;
if (tv_diff(now, &st->restart_barrier, NULL) < 0)
goto out;
if (st->clock_diff_count) { /* get status only one time */
- char *argv[] = {"audiod", "stat", "1", NULL};
- int argc = 3;
+ char *argv[] = {"audiod", "--", "stat", "-p", "-n=1", NULL};
+ int argc = 5;
PARA_INFO_LOG("clock diff count: %d\n", st->clock_diff_count);
st->clock_diff_count--;
- client_open(argc, argv, &st->ct, NULL);
+ client_open(argc, argv, &st->ct, NULL, NULL, st->btrn);
set_stat_task_restart_barrier(2);
} else {
- char *argv[] = {"audiod", "stat", NULL};
- int argc = 2;
- client_open(argc, argv, &st->ct, NULL);
+ char *argv[] = {"audiod", "--", "stat", "-p", NULL};
+ int argc = 4;
+ client_open(argc, argv, &st->ct, NULL, NULL, st->btrn);
set_stat_task_restart_barrier(5);
}
free(stat_item_values[SI_BASENAME]);
- stat_item_values[SI_BASENAME] = make_message(
- "%s: no connection to para_server\n",
- status_item_list[SI_BASENAME]);
- stat_client_write(stat_item_values[SI_BASENAME],
- SI_BASENAME);
+ stat_item_values[SI_BASENAME] = para_strdup(
+ "no connection to para_server");
+ stat_client_write_item(SI_BASENAME);
st->last_status_read = *now;
out:
- start_stop_decoders(s);
+ start_stop_decoders();
}
static void init_status_task(struct status_task *st)
{
memset(st, 0, sizeof(struct status_task));
st->task.pre_select = status_pre_select;
+ st->task.post_select = status_post_select;
st->sa_time_diff_sign = 1;
st->clock_diff_count = conf.clock_diff_count_arg;
st->current_audio_format_num = -1;
- sprintf(st->task.status, "status task");
+ sprintf(st->task.status, "stat");
+ st->btrn = btr_new_node(&(struct btr_node_description)
+ EMBRACE(.name = "stat"));
}
static void set_initial_status(void)
static void init_colors_or_die(void)
{
- int ret, i;
+ int i;
if (!want_colors())
return;
daemon_set_default_log_colors();
daemon_set_flag(DF_COLOR_LOG);
- for (i = 0; i < conf.log_color_given; i++) {
- ret = daemon_set_log_color(conf.log_color_arg[i]);
- if (ret < 0)
- exit(EXIT_FAILURE);
- }
+ for (i = 0; i < conf.log_color_given; i++)
+ daemon_set_log_color_or_die(conf.log_color_arg[i]);
}
/**
drop_privileges_or_die(conf.user_arg, conf.group_arg);
parse_config_or_die();
init_colors_or_die();
+ init_random_seed_or_die();
daemon_set_flag(DF_LOG_TIME);
daemon_set_flag(DF_LOG_HOSTNAME);
daemon_set_flag(DF_LOG_LL);
+ if (conf.log_timing_given)
+ daemon_set_flag(DF_LOG_TIMING);
if (conf.logfile_given) {
daemon_set_logfile(conf.logfile_arg);
daemon_open_log_or_die();
set_initial_status();
FOR_EACH_SLOT(i)
clear_slot(i);
- init_grabbing();
setup_signal_handling();
signal_setup_default(sig_task);
register_task(&sig_task->task);
register_task(&cmd_task->task);
register_task(&stat_task->task);
- s.default_timeout.tv_sec = 0;
- s.default_timeout.tv_usec = 99 * 1000;
+ s.default_timeout.tv_sec = 2;
+ s.default_timeout.tv_usec = 999 * 1000;
ret = schedule(&s);
PARA_EMERG_LOG("%s\n", para_strerror(-ret));
N: grab
D: grab the audio stream
L:
-U: -- grab [grab_options]
-H: grab ('splice') the audio stream at any position in the filter
-H: chain and send that data back to the client. Try
-H: para_audioc -- grab -h
-H: for the list of available options.
+U: -- grab [-m[{s|p|a}]] [-p=<parent>] [-n=<name>] [-o]
+H:
+H: grab ('splice') the audio stream at any position in the buffer
+H: tree and send that data back to the client.
+H:
+H: Options:
+H:
+H: -m Change grab mode. Defaults to sloppy grab if not given.
+H:
+H: -ms: sloppy grab
+H:
+H: -mp: pedantic grab
+H:
+H: -ma: aggressive grab
+H:
+H: The various grab modes only differ in what happens if the
+H: file descriptor to write the grabbed audio data to is not
+H: ready for writing (i.e. would block). Sloppy mode ignores
+H: the write, pedantic mode aborts and aggressive mode tries
+H: to write anyway.
+H:
+H: -p Grab output of node <parent> of the buffer tree.
+H:
+H: -n Name of the new buffer tree node. Defaults to 'grab'.
+H:
+H: -o One-shot mode: Stop grabbing if audio file changes.
---
N: help
D: display command list or help for given command
---
N: stat
D: print status information
-U: stat [item1 ...]
-H: Dump given status items (all if none given) to stdout.
+U: stat [-p] [item1 ...]
+H: Dump given status items (all if none given) to stdout. If -p is given, use
+H: parser-friendly mode.
---
N: tasks
D: list current tasks
/*
- * Copyright (C) 2006-2009 Andre Noll <maan@systemlinux.org>
+ * Copyright (C) 2006-2011 Andre Noll <maan@systemlinux.org>
*
* Licensed under the GPL v2. For licencing details see COPYING.
*/
int num_filters(int audio_format_num);
-int get_audio_format_num(char *name);
+int get_audio_format_num(const char *name);
/** enum of audio formats supported by para_audiod */
enum {AUDIOD_AUDIO_FORMATS_ENUM};
const char *name;
/** pointer to the function that handles the command */
int (*handler)(int, int, char**);
- /**
- * if the command prefers to handle the full line (rather than the usual
- * argv[] array), it stores a pointer to the corresponding line handling
- * function here. In this case, the above \a handler pointer must be NULL.
- */
- int (*line_handler)(int, char*);
/** one-line description of the command */
const char *description;
/** summary of the command line options */
/**
* Describes one instance of a receiver-filter-writer chain.
*
- * \sa receiver_node, receiver, filter, filter_node, filter_chain, writer,
- * writer_node, writer_node_group.
+ * \sa receiver_node, receiver, filter, filter_node, writer, writer_node,
+ * writer_node_group.
*/
struct slot_info {
/** Number of the audio format in this slot. */
int format;
- /** Receiver start time. */
- struct timeval rstime;
- /** Writer start time. */
- struct timeval wstime;
/** The stream_start status item announced by para_server. */
struct timeval server_stream_start;
/** The offset status item announced by para_server. */
unsigned seconds_total;
/** The receiver info associated with this slot. */
struct receiver_node *receiver_node;
- /** The active filter chain. */
- struct filter_chain *fc;
- /** The active writer node group. */
- struct writer_node_group *wng;
+ /** The array of filter nodes. */
+ struct filter_node *fns;
+ /** The array of writers attached to the last filter. */
+ struct writer_node *wns;
};
projects / paraslash.git / commitdiff
