manual: Remove outdated sentence about para_afh.

[paraslash.git] / web / manual.m4

dnl To generate the html version, execute
dnl m4 web/manual.m4 | grutatxt --toc

define(`LOCAL_LINK_NAME', `translit(`$1', `A-Z 
', `a-z__')')
define(`REMOVE_NEWLINE', `translit(`$1',`
', ` ')')

define(`REFERENCE', ./``#''`LOCAL_LINK_NAME($1)' (`REMOVE_NEWLINE($2)'))
define(`XREFERENCE', `$1' (`REMOVE_NEWLINE($2)'))
define(`EMPH', ``_''`REMOVE_NEWLINE($1)'``_'')

Paraslash user manual
=====================

This document describes how to install, configure and use the paraslash
network audio streaming system.  Most chapters start with a chapter
overview and conclude with an example section. We try to focus on
general concepts and on the interaction of the various pieces of the
paraslash package. Hence this user manual is not meant as a replacement
for the manual pages that describe all command line options of each
paraslash executable.

------------
Introduction
------------

In this chapter we give an REFERENCE(Overview, overview) of the
interactions of the two main programs contained in the paraslash
package, followed by REFERENCE(The paraslash executables, brief
descriptions) of all executables.

Overview
~~~~~~~~

The core functionality of the para suite is provided by two main
executables, para_server and para_audiod. The former maintains a
database of audio files and streams these files to para_audiod which
receives and plays the stream.

In a typical setting, both para_server and para_audiod act as
background daemons whose functionality is controlled by client
programs: the para_audioc client controls para_audiod over a local
socket while the para_client program connects to para_server over a
local or remote networking connection.

Typically, these two daemons run on different hosts but a local setup
is also possible.

A simplified picture of a typical setup is as follows
<<
<pre>
 server_host                                  client_host
 ~~~~~~~~~~~                                  ~~~~~~~~~~~
 
 +-----------+         audio stream           +-----------+
 |para_server| -----------------------------> |para_audiod|
 +-----------+                                +-----------+
      ^                                            ^
      |                                            |
      |                                            | connect
      |                                            |
      |                                            |
      |                                       +-----------+
      |                                       |para_audioc|
      |                                       +-----------+
      |
      |
      |                  connect              +-----------+
      +-------------------------------------- |para_client|
                                              +-----------+
</pre>
>>

The paraslash executables
~~~~~~~~~~~~~~~~~~~~~~~~~

*para_server*

para_server streams binary audio data (MP3, ...) over local and/or
remote networks. It listens on a TCP port and accepts commands such
as play, stop, pause, next from authenticated clients. There are
many more commands though, see the man page of para_server for a
description of all commands.

It supports three built-in network streaming protocols
(senders/receivers): HTTP, DCCP, or UDP. This is explained in more
detail in the section on REFERENCE(Networking, networking).

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.

Additional information may be added to the database to allow
fine-grained selection based on various properties of the audio file,
including information found in (ID3) tags. However, old-fashioned
playlists are also supported.

It is also possible to store images (album covers) and lyrics in the
database and associate these to the corresponding audio files.

The section on the REFERENCE(The audio file selector, audio file
selector) discusses this topic.


*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
with a symmetric RC4 session key. For each user of paraslash you must
create a public/secret RSA key pair for authentication.

If para_client is started without non-option arguments, an interactive
session (shell) is started. Command history and command completion are
supported through libreadline.

*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.

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 of the decoding process. Like para_client, para_audioc supports
interactive sessions on systems with libreadline.

*para_recv*

A command line HTTP/DCCP/UDP stream grabber. The http mode is
compatible with arbitrary HTTP streaming sources (e.g. icecast).
In addition to the three network streaming modes, para_recv can also
operate in local (afh) mode. In this mode it writes the content of
an audio file on the local file system in complete chunks to stdout,
optionally 'just in time'. This allows to cut an audio file without
first decoding it, and it enables third-party software which is unaware
of the particular audio format to send complete frames in real time.

*para_filter*

A filter program that reads from STDIN and writes to STDOUT.
Like para_recv, this is an atomic building block which can be used to
assemble higher-level audio receiving facilities. It combines several
different functionalities in one tool: decoders for multiple audio
formats and a number of processing filters, among these a normalizer
for audio volume.

*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.

*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_play*

A command line audio player.

*para_gui*

Curses-based gui that presents status information obtained in a curses
window. Appearance can be customized via themes. para_gui provides
key-bindings for the most common server commands and new key-bindings
can be added easily.


*para_fade*

An alarm clock and volume-fader for OSS and ALSA.

-----------
Quick start
-----------

This chapter lists the REFERENCE(Requirements, necessary software)
that must be installed to compile the paraslash package, describes
how to REFERENCE(Installation, compile and install) the paraslash
source code and the steps that have to be performed in order to
REFERENCE(Quick start, set up) a typical server and client.

Requirements
~~~~~~~~~~~~
For the impatient:

        git clone git://git.tuebingen.mpg.de/osl
        cd osl && make && sudo make install && sudo ldconfig
        sudo apt-get install autoconf libssl-dev help2man gengetopt \
              libmad0-dev libid3tag0-dev libasound2-dev libvorbis-dev \
              libfaad-dev libspeex-dev libFLAC-dev libsamplerate-dev \
              libasound2-dev libao-dev libreadline-dev libncurses-dev

Detailed description: In any case you'll need

        - XREFERENCE(http://systemlinux.org/~maan/osl/, libosl).
        The _object storage layer_ library is used by para_server. To
        clone the source code repository, execute

        git clone git://git.tuebingen.mpg.de/osl

        - XREFERENCE(ftp://ftp.gnu.org/pub/gnu/gcc, gcc) or
        XREFERENCE(http://clang.llvm.org, clang). All gcc versions
        >= 3.3 are currently supported. Clang version 1.1 or newer
        should work as well.

        - XREFERENCE(ftp://ftp.gnu.org/pub/gnu/make, gnu make) is
        also shipped with the disto. On BSD systems the gnu make
        executable is often called gmake.

        - XREFERENCE(ftp://ftp.gnu.org/pub/gnu/bash, bash). Some
        scripts which run during compilation require the EMPH(Bourne
        again shell).  It is most likely already installed.

        - XREFERENCE(ftp://ftp.gnu.org/pub/gnu/gengetopt/, gengetopt)
        is needed to generate the C code for the command line parsers
        of all paraslash executables.

        - XREFERENCE(ftp://ftp.gnu.org/pub/gnu/help2man, help2man)
        is used to create the man pages.

Optional:

        - XREFERENCE(http://www.openssl.org/, openssl) or
        XREFERENCE(ftp://ftp.gnupg.org/gcrypt/libgcrypt/, libgcrypt).
        At least one of these two libraries is needed as the backend
        for cryptographic routines on both the server and the client
        side. Both openssl and libgcrypt are usually shipped with the
        distro, but you might have to install the development package
        (libssl-dev or libgcrypt-dev on debian systems) as well.

        - XREFERENCE(http://www.underbit.com/products/mad/, libmad).
        To compile in MP3 support for paraslash, the development
        package must be installed. It is called libmad0-dev on
        debian-based systems. Note that libmad is not necessary on
        the server side, i.e. for sending MP3 files.

        - XREFERENCE(http://www.underbit.com/products/mad/,
        libid3tag). For version-2 ID3 tag support, you'll need
        the libid3tag development package libid3tag0-dev. Without
        libid3tag, only version one tags are recognized.

        - XREFERENCE(http://www.xiph.org/downloads/, ogg vorbis).
        For ogg vorbis streams you'll need libogg, libvorbis,
        libvorbisfile. The corresponding Debian packages are called
        libogg-dev and libvorbis-dev.

        - XREFERENCE(http://www.audiocoding.com/, libfaad).  For aac
        files (m4a) you'll need libfaad (libfaad-dev).

        - XREFERENCE(http://www.speex.org/, speex). In order to stream
        or decode speex files, libspeex (libspeex-dev) is required.

        - XREFERENCE(http://flac.sourceforge.net/, flac). To stream
        or decode files encoded with the _Free Lossless Audio Codec_,
        libFLAC (libFLAC-dev) must be installed.

        - XREFERENCE(http://www.mega-nerd.com/SRC/index.html,
        libsamplerate). The resample filter will only be compiled if
        this library is installed. Debian package: libsamplerate-dev.

        - XREFERENCE(ftp://ftp.alsa-project.org/pub/lib/, alsa-lib). On
        Linux, you'll need to have ALSA's development package
        libasound2-dev installed.

        - XREFERENCE(http://downloads.xiph.org/releases/ao/,
        libao). Needed to build the ao writer (ESD, PulseAudio,...).
        Debian package: libao-dev.

        - XREFERENCE(ftp://ftp.gnu.org/pub/gnu/ncurses, curses). Needed
        for para_gui. Debian package: libncurses-dev.

        - XREFERENCE(http://cnswww.cns.cwru.edu/php/chet/readline/rltop.html,
        GNU Readline). If this library (libreadline-dev) is installed,
        para_client, para_audioc and para_play support interactive
        sessions.

Installation
~~~~~~~~~~~~

First make sure all non-optional packages listed in the section on
REFERENCE(Requirements, required software) are installed on your
system.

You don't need everything listed there. In particular, MP3, OGG/Vorbis,
OGG/Speex 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 decoder library (not even the MP3 decoding library
libmad) is needed for para_server if you only want to stream MP3 or WMA
files. Also, it's fine to use para_server on a box without sound card.

Next, install the paraslash package on all machines, you'd like this
software to run on. If you compile from a released tarball, execute

        (./configure && make) > /dev/null

When compiling from git or from snapshots downloaded via gitweb,
the above command will not work because the configure script is not
included in the git repository. In this case the following command
should be used instead:

        ./autogen.sh

This runs autoconf to generate the configure script, then runs it as
above. Therefore you'll need autoconf for this to work.

There should be no errors but probably some warnings about missing
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 (optionally)

        make test

to run the paraslash test suite. If all tests pass, execute as root

        make install

to install executables under /usr/local/bin and the man pages under
/usr/local/man.

Configuration
~~~~~~~~~~~~~

*Step 1*: Create a paraslash user

In order to control para_server at runtime you must create a paraslash
user. As authentication is based on the RSA crypto system you'll have
to create an RSA key pair. If you already have a user and an RSA key
pair, you may skip this step.

In this section we'll assume a typical setup: You would like to run
para_server on some host called server_host as user foo, and you want
to connect to para_server from another machine called 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/id_rsa.pub.$user
        perms=AFS_READ,AFS_WRITE,VSS_READ,VSS_WRITE
        mkdir -p ~/.paraslash
        echo "user $user $key $perms" >> $target

Next, change to the "bar" account on client_host and generate the
key pair with the commands

        ssh-keygen -t rsa -b 2048
        # hit enter twice to create a key with no passphrase

This generates the two files id_rsa and id_rsa.pub in ~/.ssh. Note
that paraslash can also read keys generated by the "openssl genrsa"
command. However, since keys created with ssh-keygen can also be used
for ssh, this method is recommended.

Note that para_server refuses to use a key if it is shorter than 2048
bits. In particular, the RSA keys of paraslash 0.3.x will not work
with version 0.4.x. Moreover, para_client refuses to use a (private)
key which is world-readable.

para_server only needs to know the public key of the key pair just
created. Copy this public key to server_host:

        src=~/.ssh/id_rsa.pub
        dest=.paraslash/id_rsa.pub.$LOGNAME
        scp $src foo@server_host:$dest

Finally, tell para_client to connect to server_host:

        conf=~/.paraslash/client.conf
        echo 'hostname server_host' > $conf


*Step 2*: Start para_server

Before starting the server make sure you have write permissions to
the directory /var/paraslash that has been created during installation:

       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@client_host 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.

*Step 3*: Create and populate the database

An empty database is created with

        para_client init

This initializes a couple of empty tables under
~/.paraslash/afs_database-0.4. 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-0.4

in case something went wrong.

Next, you need to add some audio files to that database so that
para_server knows about them. 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 files. Note that the table only contains data
about the audio files found, not the files themselves.

You may print the list of all known audio files with

        para_client ls

*Step 4*: Configure para_audiod

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

In order to make this directory writable for para_audiod, execute
as bar@client_host

       sudo chown $LOGNAME /var/paraslash


We will also have to tell para_audiod that it should receive the
audio stream from server_host via http:

        para_audiod -l info -r '.:http -i server_host'

You should now be able to listen to the audio stream once para_server
starts streaming. To activate streaming, execute

        para_client play

Since no playlist has been specified yet, the "dummy" mode which
selects all known audio files is activated automatically. See the
section on the REFERENCE(The audio file selector, audio file selector)
for how to use playlists and moods to specify which files should be
streamed in which order.

*Troubleshooting*

It did not work? To find out why, try to receive, decode and play the
stream manually using para_recv, para_filter and para_write as follows.

For simplicity we assume that you're running Linux/ALSA and that only
MP3 files have been added to the database.

        para_recv -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

Double check what is logged by para_server and use the --loglevel
option of para_recv, para_filter and para_write to increase verbosity.

---------------
User management
---------------

para_server uses a challenge-response mechanism to authenticate
requests from incoming connections, similar to ssh's public key
authentication method. Authenticated connections are encrypted using
the RC4 stream cipher.

In this chapter we briefly describe RSA and RC4 and sketch the
REFERENCE(Client-server authentication, authentication handshake)
between para_client and para_server. User management is discussed
in the section on REFERENCE(The user_list file, the user_list file).
These sections are all about communication between the client and the
server. Connecting para_audiod is a different matter and is described
in a REFERENCE(Connecting para_audiod, separate section).



RSA and RC4
~~~~~~~~~~~

RSA is an asymmetric block cipher which is used in many applications,
including ssh and gpg. An RSA key consists in fact of two keys,
called the public key and the private key. A message can be encrypted
with either key and only the counterpart of that key can decrypt
the message. While RSA can be used for both signing and encrypting
a message, paraslash uses RSA only for the latter purpose. The
RSA public key encryption and signatures algorithms are defined in
detail in RFC 2437.

RC4 is a stream cipher, i.e. the input is XORed with a pseudo-random
key stream to produce the output. Decryption uses the same function
calls as encryption. While RC4 supports variable key lengths,
paraslash uses a fixed length of 256 bits, which is considered a
strong encryption by today's standards. Since the same key must never
be used twice, a different, randomly-generated key is used for every
new connection.

Client-server authentication
~~~~~~~~~~~~~~~~~~~~~~~~~~~~

The authentication handshake between para_client and para_server goes
as follows:

        - para_client connects to para_server and sends an
        authentication request for a user. It does so by connecting
        to TCP port 2990 of the server host. This port is called the
        para_server _control port_.

        - para_server accepts the connection and forks a child process
        which handles the incoming request. The parent process keeps
        listening on the control port while the child process (also
        called para_server below) continues as follows.

        - para_server loads the RSA public key of that user, fills a
        fixed-length buffer with random bytes, encrypts that buffer
        using the public key and sends the encrypted buffer to the
        client. The first part of the buffer is the challenge which
        is used for authentication while the second part is the RC4
        session key.

        - para_client receives the encrypted buffer and decrypts it
        with the user's private key, thereby obtaining the challenge
        buffer and the session key. It sends the SHA1 hash value of
        the challenge back to para_server and stores the session key
        for further use.

        - para_server also computes the SHA1 hash of the challenge
        and compares it against what was sent back by the client.

        - If the two hashes do not match, the authentication has
        failed and para_server closes the connection.

        - Otherwise the user is considered authenticated and the client
        is allowed to proceed by sending a command to be executed. From
        this point on the communication is encrypted using the RC4
        stream cipher with the session key known to both peers.

paraslash relies on the quality of the pseudo-random bytes provided
by the crypto library (openssl or libgcrypt), on the security of
the implementation of the RSA and RC4 crypto routines and on the
infeasibility to invert the SHA1 function.

Neither para_server or para_client create RSA keys on their own. This
has to be done once for each user as sketched in REFERENCE(Quick start,
Quick start) and discussed in more detail REFERENCE(The user_list
file, below).

The user_list file
~~~~~~~~~~~~~~~~~~

At startup para_server reads the user list file which contains one
line per user. The default location of the user list file may be
changed with the --user-list option.

There should be at least one user in this file. Each user must have
an RSA key pair. The public part of the key is needed by para_server
while the private key is needed by para_client. Each line of the
user list file must be of the form

        user <username> <key> <perms>

where _username_ is an arbitrary string (usually the user's login
name), _key_ is the full path to that user's public RSA key, and
_perms_ is a comma-separated list of zero or more of the following
permission bits:

        +---------------------------------------------------------+
        | AFS_READ  | read the contents of the databases          |
        +-----------+---------------------------------------------+
        | AFS_WRITE | change database contents                    |
        +-----------+---------------------------------------------+
        | VSS_READ  | obtain information about the current stream |
        +-----------+---------------------------------------------+
        | VSS_WRITE | change the current stream                   |
        +---------------------------------------------------------+

The permission bits specify which commands the user is allowed to
execute. The output of

        para_client help

contains in the third column the permissions needed to execute the
command.

It is possible to make para_server reread the user_list file by
executing the paraslash "hup" command or by sending SIGHUP to the
PID of para_server.


Connecting para_audiod
~~~~~~~~~~~~~~~~~~~~~~

para_audiod listens on a Unix domain socket. Those sockets are
for local communication only, so only local users can connect to
para_audiod. The default is to let any user connect but this can be
restricted on platforms that support UNIX socket credentials which
allow para_audiod to obtain the Unix credentials of the connecting
process.

Use para_audiod's --user-allow option to allow connections only for
a limited set of users.

-----------------------
The audio file selector
-----------------------

paraslash comes with a sophisticated audio file selector (AFS),
whose main task is to determine which file to stream next, based on
information on the audio files stored in a database. It communicates
also with para_client whenever an AFS command is executed, for example
to answer a database query.

Besides the traditional playlists, AFS supports audio file selection
based on _moods_ which act as a filter that limits the set of all
known audio files to those which satisfy certain criteria.  It also
maintains tables containing images (e.g. album cover art) and lyrics
that can be associated with one or more audio files.

AFS uses XREFERENCE(http://systemlinux.org/~maan/osl/, libosl), the
object storage layer library, as the backend library for storing
information on audio files, playlists, etc. This library offers
functionality similar to a relational database, but is much more
lightweight than a full database backend.

In this chapter we sketch the setup of the REFERENCE(The AFS process,
AFS process) during server startup and proceed with the description
of the REFERENCE(Database layout, layout) of the various database
tables. The section on REFERENCE(Playlists and moods, playlists
and moods) explains these two audio file selection mechanisms
in detail and contains pratical examples. The way REFERENCE(File
renames and content changes, file renames and content changes) are
detected is discussed briefly before the REFERENCE(Troubleshooting,
Troubleshooting) section concludes the chapter.

The AFS process
~~~~~~~~~~~~~~~

On startup, para_server forks to create the AFS process which opens
the OSL database tables. The server process communicates with the
AFS process via pipes and shared memory. Usually, the AFS process
awakes only briefly whenever the current audio file changes. The AFS
process determines the next audio file, opens it, verifies it has
not been changed since it was added to the database and passes the
open file descriptor to the server process, along with audio file
meta-data such as file name, duration, audio format and so on. The
server process then starts to stream the audio file.

The AFS process also accepts connections from local clients via
a well-known socket. However, only child processes of para_server
may connect through this socket. All server commands that have the
AFS_READ or AFS_WRITE permission bits use this mechanism to query or
change the database.

Database layout
~~~~~~~~~~~~~~~

*The audio file table*

This is the most important and usually also the largest table of the
AFS database. It contains the information needed to stream each audio
file. In particular the following data is stored for each audio file.

        - SHA1 hash value of the audio file contents. This is computed
        once when the file is added to the database. Whenever AFS
        selects this audio file for streaming the hash value is
        recomputed and checked against the value stored in the
        database to detect content changes.

        - The time when this audio file was last played.

        - The number of times the file has been played so far.

        - The attribute bitmask.

        - The image id which describes the image associated with this
        audio file.

        - The lyrics id which describes the lyrics associated with
        this audio file.

        - The audio format id (MP3, OGG, ...).

        - An amplification value that can be used by the amplification
        filter to pre-amplify the decoded audio stream.

        - The chunk table. It describes the location and the timing
        of the building blocks of the audio file. This is used by
        para_server to send chunks of the file at appropriate times.

        - The duration of the audio file.

        - Tag information contained in the audio file (ID3 tags,
        Vorbis comments, ...).

        - The number of channels

        - The encoding bitrate.

        - The sampling frequency.

To add or refresh the data contained in the audio file table, the _add_
command is used. It takes the full path of either an audio file or a
directory. In the latter case, the directory is traversed recursively
and all files which are recognized as valid audio files are added to
the database.

*The attribute table*

The attribute table contains two columns, _name_ and _bitnum_. An
attribute is simply a name for a certain bit number in the attribute
bitmask of the audio file table.

Each of the 64 bits of the attribute bitmask can be set for each
audio file individually. Hence up to 64  different attributes may be
defined. For example, "pop", "rock", "blues", "jazz", "instrumental",
"german_lyrics", "speech", whatever. You are free to choose as
many attributes as you like and there are no naming restrictions
for attributes.

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 this pattern:

        para_client setatt test+ '/test/directory/*'

The command

        para_client -- ls -lv

gives you a verbose listing of your audio files also showing 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.

*Blob tables*

The image, lyrics, moods and playlists tables are all blob tables.
Blob tables consist of three columns each: The identifier which is
a positive non-negative number that is auto-incremented, the name
(an arbitrary string) and the content (the blob).

All blob tables support the same set of actions: cat, ls, mv, rm
and add. Of course, _add_ is used for adding new blobs to the table
while the other actions have the same meaning as the corresponding
Unix commands. The paraslash commands to perform these actions are
constructed as the concatenation of the table name and the action. For
example addimg, catimg, lsimg, mvimg, rmimg are the commands that
manipulate or query the image table.

The add variant of these commands is special as these commands read
the blob contents from stdin. To add an image to the image table the
command

        para addimg image_name < file.jpg

can be used.

Note that the images and lyrics are not interpreted at all, and also
the playlist and the mood blobs are only investigated when the mood
or playlist is activated with the select command.

*The score table*

Unlike all other tables the contents of the score table remain in
memory and are never stored on disk. The score table contains two
columns: The SHA1 hash value (of an audio file) and its current
score.

However, only those files which are admissible for the current mood
or playlist are contained in the score table. The audio file selector
always chooses the row with the highest score as the file to stream
next. While doing so, it computes the new score and updates the
last_played and the num_played fields in the audio file table.

The score table is recomputed by the select command which loads a
mood or playlist. Audio files are chosen for streaming from the rows
of the score table on a highest-score-first basis.


Playlists and moods
~~~~~~~~~~~~~~~~~~~

Playlists and moods offer two different ways of specifying the set of
admissible files. A playlist in itself describes a set of admissible
files. A mood, in contrast, describes the set of admissible files in
terms of attributes and other type of information available in the
audio file table. As an example, a mood can define a filename pattern,
which is then matched against the names of audio files in the table.

*Playlists*

Playlists are accommodated in the playlist table of the afs database,
using the aforementioned blob format for tables. A new playlist is
created with the addpl command by specifying the full (absolute)
paths of all desired audio files, separated by newlines. Example:

        find /my/mp3/dir -name "*.mp3" | para addpl my_playlist

If _my_playlist_ already exists it is overwritten. To activate the
new playlist, execute

        para select p/my_playlist

The audio file selector will assign scores to each entry of the list,
in descending order so that files will be selected in order. If a
file could not be opened for streaming, its entry is removed from
the score table (but not from the playlist).

*Moods*

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.

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 only syntactic
sugar 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.

        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.

        artist_matches <pattern>
        album_matches <pattern>
        title_matches <pattern>
        comment_matches <pattern>

Takes an extended regular expression and matches iff the text of the
corresponding tag of the audio file matches the pattern. If the tag
is not set, the empty string is matched against the pattern.

        year ~ <num>
        bitrate ~ <num>
        frequency ~ <num>
        channels ~ <num>
        num_played ~ <num>

Takes a comparator ~ of the set {<, =, <=, >, >=, !=} and a number
<num>. Matches an audio file iff the condition <val> ~ <num> is
satisfied where val is the corresponding value of the audio file
(value of the year tag, bitrate in kbit/s, frequency in Hz, channel
count, play count).

The year tag is special as its value is undefined if the audio file
has no year tag or the content of the year tag is not a number. Such
audio files never match. Another difference is the special treatment
if the year tag is a two-digit number. In this case either 1900 or
2000 is added to the tag value, depending on whether the number is
greater than 2000 plus the current year.


*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



File renames and content changes
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

Since the audio file selector knows the SHA1 of each audio file that
has been added to the afs database, it recognizes if the content of
a file has changed, e.g. because an ID3 tag was added or modified.
Also, if a file has been renamed or moved to a different location,
afs will detect that an entry with the same hash value already exists
in the audio file table.

In both cases it is enough to just re-add the new file. In the
first case (file content changed), the audio table is updated, while
metadata such as the num_played and last_played fields, as well as
the attributes, remain unchanged. In the other case, when the file
is moved or renamed, only the path information is updated, all other
data remains as before.

It is possible to change the behaviour of the add command by using the
"-l" (lazy add) or the "-f" (force add) option.

Troubleshooting
~~~~~~~~~~~~~~~

Use the debug loglevel (-l debug) to show debugging info. All paraslash
executables have a brief online help which is displayed when -h is
given. The --detailed-help option prints the full help text.

If para_server crashed or was killed by SIGKILL (signal 9), it
may refuse to start again because of "dirty osl tables". In this
case you'll have to run the oslfsck program of libosl to fix your
database:

        oslfsck -fd ~/.paraslash/afs_database-0.4

However, make sure para_server isn't running before executing oslfsck.

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-0.4

Be aware that this removes all attribute definitions, all playlists
and all mood definitions and requires to re-initialize the tables.

Although oslfsck fixes inconsistencies in database tables it doesn't
care about the table contents. To check for invalid table contents, use

        para_client check

This prints out references to missing audio files as well as invalid
playlists and mood definitions.

Similarly, para_audiod refuses to start if its socket file exists, since
this indicates that another instance of para_audiod is running. After
a crash a stale socket file might remain and you must run

        para_audiod --force

once to fix it up.

---------------------------------------
Audio formats and audio format handlers
---------------------------------------

Audio formats
~~~~~~~~~~~~~

The following audio formats are supported by paraslash:

*MP3*

Mp3, MPEG-1 Audio Layer 3, is a common audio format for audio storage,
designed as part of its MPEG-1 standard.  An MP3 file is made up of
multiple MP3 frames, which consist of a header and a data block. The
size of an MP3 frame depends on the bit rate and on the number
of channels. For a typical CD-audio file (sample rate of 44.1 kHz
stereo), encoded with a bit rate of 128 kbit, an MP3 frame is about
400 bytes large.

*OGG/Vorbis*

OGG is a standardized audio container format, while Vorbis is an
open source codec for lossy audio compression. Since Vorbis is most
commonly made available via the OGG container format, it is often
referred to as OGG/Vorbis. The OGG container format divides data into
chunks called OGG pages. A typical OGG page is about 4KB large. The
Vorbis codec creates variable-bitrate (VBR) data, where the bitrate
may vary considerably.

*OGG/Speex*

Speex is an open-source speech codec that is based on CELP (Code
Excited Linear Prediction) coding. It is designed for voice
over IP applications, has modest complexity and a small memory
footprint. Wideband and narrowband (telephone quality) speech are
supported. As for Vorbis audio, Speex bit-streams are often stored
in OGG files. As of 2012 this codec is considered obsolete since the
Oppus codec, described below, surpasses its performance in all areas.

*OGG/Opus*

Opus is a lossy audio compression format standardized through RFC
6716 in 2012. It combines the speech-oriented SILK codec and the
low-latency CELT (Constrained Energy Lapped Transform) codec. Like
OGG/Vorbis and OGG/Speex, Opus data is usually encapsulated in OGG
containers. All known software patents which cover Opus are licensed
under royalty-free terms.

*AAC*

Advanced Audio Coding (AAC) is a standardized, lossy compression
and encoding scheme for digital audio which is the default audio
format for Apple's iPhone, iPod, iTunes. Usually MPEG-4 is used as
the container format and audio files encoded with AAC have the .m4a
extension. A typical AAC frame is about 700 bytes large.

*WMA*

Windows Media Audio (WMA) is an audio data compression technology
developed by Microsoft. A WMA file is usually encapsulated in the
Advanced Systems Format (ASF) container format, which also specifies
how meta data about the file is to be encoded. The bit stream of WMA
is composed of superframes, each containing one or more frames of
2048 samples. For 16 bit stereo a WMA superframe is about 8K large.

*FLAC*

The Free Lossless Audio Codec (FLAC) compresses audio without quality
loss. It gives better compression ratios than a general purpose
compressor like zip or bzip2 because FLAC is designed specifically
for audio. A FLAC-encoded file consits of frames of varying size, up
to 16K. Each frame starts with a header that contains all information
necessary to decode the frame.

Meta data
~~~~~~~~~

Unfortunately, each audio format has its own conventions how meta
data is added as tags to the audio file.

For MP3 files, ID3, version 1 and 2 are widely used. ID3 version 1
is rather simple but also very limited as it supports only artist,
title, album, year and comment tags. Each of these can only be at most
32 characters long. ID3, version 2 is much more flexible but requires
a separate library being installed for paraslash to support it.

Ogg vorbis, ogg speex and flac files contain meta data as Vorbis
comments, which are typically implemented as strings of the form
"[TAG]=[VALUE]". Unlike ID3 version 1 tags, one may use whichever
tags are appropriate for the content.

AAC files usually use the MPEG-4 container format for storing meta
data while WMA files wrap meta data as special objects within the
ASF container format.

paraslash only tracks the most common tags that are supported by
all tag variants: artist, title, year, album, comment. When a file
is added to the AFS database, the meta data of the file is extracted
and stored in the audio file table.

Chunks and chunk tables
~~~~~~~~~~~~~~~~~~~~~~~

paraslash uses the word "chunk" as common term for the building blocks
of an audio file. For MP3 files, a chunk is the same as an MP3 frame,
while for OGG files a chunk is an OGG page, etc.  Therefore the chunk
size varies considerably between audio formats, from a few hundred
bytes (MP3) up to 16K (FLAC).

The chunk table contains the offsets within the audio file that
correspond to the chunk boundaries of the file. Like the meta data,
the chunk table is computed and stored in the database whenever an
audio file is added.

The paraslash senders (see below) always send complete chunks. The
granularity for seeking is therefore determined by the chunk size.

Audio format handlers
~~~~~~~~~~~~~~~~~~~~~

For each audio format paraslash contains an audio format handler whose
first task is to tell whether a given file is a valid audio file of
this type. If so, the audio file handler extracts some technical data
(duration, sampling rate, number of channels etc.), computes the
chunk table and reads the meta data.

The audio format handler code is linked into para_server and executed
via the _add_ command. The same code is also available as a stand-alone
tool, para_afh, which can be used to print the technical data, the
chunk table and the meta data of a file.

----------
Networking
----------

Paraslash uses different network connections for control and data.
para_client communicates with para_server over a dedicated TCP control
connection. To transport audio data, separate data connections are
used. For these data connections, a variety of transports (UDP, DCCP,
HTTP) can be chosen.

The chapter starts with the REFERENCE(The paraslash control
service, control service), followed by a section on the various
REFERENCE(Streaming protocols, streaming protocols) in which the data
connections are described. The way audio file headers are embedded into
the stream is discussed REFERENCE(Streams with headers and headerless
streams, briefly) before the REFERENCE(Networking examples, example
section) which illustrates typical commands for real-life scenarios.

Both IPv4 and IPv6 are supported.

The paraslash control service
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

para_server is controlled at runtime via the paraslash control
connection. This connection is used for server commands (play, stop,
...) as well as for afs commands (ls, select, ...).

The server listens on a TCP port and accepts connections from clients
that connect the open port. Each connection causes the server to fork
off a client process which inherits the connection and deals with that
client only. In this classical accept/fork approach the server process
is unaffected if the child dies or goes crazy for whatever reason. In
fact, the child process can not change address space of server process.

The section on REFERENCE(Client-server authentication, client-server
authentication) above described the early connection establishment
from the crypto point of view. Here it is described what happens
after the connection (including crypto setup) has been established.
There are four processes involved during command dispatch as sketched
in the following diagram.

<<
<pre>
 server_host                                   client_host
 ~~~~~~~~~~~                                   ~~~~~~~~~~~

 +-----------+             connect            +-----------+
 |para_server|<------------------------------ |para_client|
 +-----------+                                +-----------+
      |                                             ^
      |     fork   +---+                            |
      +----------> |AFS|                            |
      |            +---+                            |
      |              ^                              |
      |              |                              |
      |              | connect (cookie)             |
      |              |                              |
      |              |                              |
      |    fork   +-----+    inherited connection   |
      +---------->|child|<--------------------------+
                  +-----+
</pre>
>>

Note that the child process is not a child of the afs process,
so communication of these two processes has to happen via local
sockets. In order to avoid abuse of the local socket by unrelated
processes, a magic cookie is created once at server startup time just
before the server process forks off the AFS process. This cookie is
known to the server, AFS and the child, but not to unrelated processes.

There are two different kinds of commands: First there are commands
that cause the server to respond with some answer such as the list
of all audio files. All but the addblob commands (addimg, addlyr,
addpl, addmood) are of this kind. The addblob commands add contents
to the database, so they need to transfer data the other way round,
from the client to the server.

There is no knowledge about the server commands built into para_client,
so it does not know about addblob commands. Instead, it inspects the
first data package sent by the server for a magic string. If this
string was found, it sends STDIN to the server, otherwise it dumps
data from the server to STDOUT.

Streaming protocols
~~~~~~~~~~~~~~~~~~~

A network (audio) stream usually consists of one streaming source,
the _sender_, and one or more _receivers_ which read data over the
network from the streaming source.

Senders are thus part of para_server while receivers are part of
para_audiod. Moreover, there is the stand-alone tool para_recv which
can be used to manually download a stream, either from para_server
or from a web-based audio streaming service.

The following three streaming protocols are supported by paraslash:

        - HTTP. Recommended for public streams that can be played by
        any player like mpg123, xmms, itunes, winamp, etc. The HTTP
        sender is supported on all operating systems and all platforms.

        - DCCP. Recommended for LAN streaming. DCCP is currently
        available only for Linux.

        - UDP. Recommended for multicast LAN streaming.

See the Appendix on REFERENCE(Network protocols, network protocols)
for brief descriptions of the various protocols relevant for network
audio streaming with paraslash.

It is possible to activate more than one sender simultaneously.
Senders can be controlled at run time and via config file and command
line options.

Note that audio connections are _not_ encrypted. Transport or Internet
layer encryption should be used if encrypted data connections are
needed.

Since DCCP and TCP are both connection-oriented protocols, connection
establishment/teardown and access control are very similar between
these two streaming protocols. UDP is the most lightweight option,
since in contrast to TCP/DCCP it is connectionless. It is also the
only protocol supporting IP multicast.

The HTTP and the DCCP sender listen on a (TCP/DCCP) port waiting for
clients to connect and establish a connection via some protocol-defined
handshake mechanism. Both senders maintain two linked lists each:
The list of all clients which are currently connected, and the list
of access control entries which determines who is allowed to connect.
IP-based access control may be configured through config file and
command line options and via the "allow" and "deny" sender subcommands.

Upon receiving a GET request from the client, the HTTP sender sends
back a status line and a message. The body of this message is the
audio stream. This is common practice and is supported by many popular
clients which can thus be used to play a stream offered by para_server.
For DCCP things are a bit simpler: No messages are exchanged between
the receiver and sender. The client simply connects and the sender
starts to stream.

DCCP is an experimental protocol which offers a number of new features
not available for TCP. Both ends can negotiate these features using
a built-in negotiation mechanism. In contrast to TCP/HTTP, DCCP is
datagram-based (no retransmissions) and thus should not be used over
lossy media (e.g. WiFi networks). One useful feature offered by DCCP
is access to a variety of different congestion-control mechanisms
called CCIDs. Two different CCIDs are available per default on Linux:


        - _CCID 2_. A Congestion Control mechanism similar to that
        of TCP. The sender maintains a congestion window and halves
        this window in response to congestion.


        - _CCID-3_. Designed to be fair when competing for bandwidth.
        It has lower variation of throughput over time compared with
        TCP, which makes it suitable for streaming media.

Unlike the HTTP and DCCP senders, the UDP sender maintains only a
single list, the _target list_. This list describes the set of clients
to which the stream is sent. There is no list for access control and
no "allow" and "deny" commands for the UDP sender. Instead, the "add"
and "delete" commands can be used to modify the target list.

Since both UDP and DCCP offer an unreliable datagram-based transport,
additional measures are necessary to guard against disruptions over
networks that are lossy or which may be subject to interference (as
is for instance the case with WiFi). Paraslash uses FEC (Forward
Error Correction) to guard against packet losses and reordering. The
stream is FEC-encoded before it is sent through the UDP socket and
must be decoded accordingly on the receiver side.

The packet size and the amount of redundancy introduced by FEC can
be configured via the FEC parameters which are dictated by server
and may also be configured through the "sender" command.  The FEC
parameters are encoded in the header of each network packet, so no
configuration is necessary on the receiver side. See the section on
REFERENCE(Forward error correction, FEC) below.

Streams with headers and headerless streams
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

For OGG/Vorbis, OGG/Speex and wma streams, some of the information
needed to decode the stream is only contained in the audio file
header of the container format but not in each data chunk. Clients
must be able to obtain this information in case streaming starts in
the middle of the file or if para_audiod is started while para_server
is already sending a stream.

This is accomplished in different ways, depending on the streaming
protocol. For connection-oriented streams (HTTP, DCCP) the audio file
header is sent prior to audio file data. This technique however does
not work for the connectionless UDP transport. Hence the audio file
header is periodically being embedded into the UDP audio data stream.
By default, the header is resent after five seconds. The receiver has
to wait until the next header arrives before it can start decoding
the stream.

Examples
~~~~~~~~

The sender command of para_server allows to (de-)activate senders
and to change the access permissions senders at runtime. The "si"
(server info) command is used to list the streaming options of the
currently running server as well as the various sender access lists.

-> Show client/target/access lists:

        para_client si

-> Obtain general help for the sender command:

        para_client help sender

-> Get help for a specific sender (contains further examples):

        s=http # or dccp or udp
        para_client sender $s help

By default para_server activates both the HTTP and th DCCP sender on
startup. This can be changed via command line options or para_server's
config file.

-> List config file options for senders:

        para_server -h

All senders share the "on" and "off" commands, so senders may be
activated and deactivated independently of each other.

-> Switch off the http sender:

        para_client sender http off

-> Receive a DCCP stream using CCID2 and write the output into a file:

        host=foo.org; ccid=2; filename=bar
        para_recv --receiver "dccp --host $host --ccid $ccid" > $filename

Note the quotes around the arguments for the dccp receiver. Each
receiver has its own set of command line options and its own command
line parser, so arguments for the dccp receiver must be protected
from being interpreted by para_recv.

-> Start UDP multicast, using the default multicast address:

        para_client sender udp add 224.0.1.38

-> Receive FEC-encoded multicast stream and write the output into a file:

        filename=foo
        para_recv -r udp > $filename

-> Add an UDP unicast for a client to the target list of the UDP sender:

        t=client.foo.org
        para_client sender udp add $t

-> Receive this (FEC-encoded) unicast stream:

        filename=foo
        para_recv -r 'udp -i 0.0.0.0' > $filename

-> Create a minimal config for para_audiod for HTTP streams:

        c=$HOME/.paraslash/audiod.conf.min; s=server.foo.com
        echo receiver \".:http -i $s\" > $c
        para_audiod --config $c

-------
Filters
-------

A paraslash filter is a module which transforms an input stream into
an output stream. Filters are included in the para_audiod executable
and in the stand-alone tool para_filter which usually contains the
same modules.

While para_filter reads its input stream from STDIN and writes
the output to STDOUT, the filter modules of para_audiod are always
connected to a receiver which produces the input stream and a writer
which absorbs the output stream.

Some filters depend on a specific library and are not compiled in
if this library was not found at compile time. To see the list of
supported filters, run para_filter and para_audiod with the --help
option. The output looks similar to the following:

        Available filters:
                compress wav amp fecdec wmadec prebuffer oggdec aacdec mp3dec

Out of these filter modules, a chain of filters can be constructed,
much in the way Unix pipes can be chained, and analogous to the use
of modules in gstreamer: The output of the first filter becomes the
input of the second filter. There is no limitation on the number of
filters and the same filter may occur more than once.

Like receivers, each filter has its own command line options which
must be quoted to protect them from the command line options of
the driving application (para_audiod or para_filter). Example:

        para_filter -f 'mp3dec --ignore-crc' -f 'compress --damp 1'

For para_audiod, each audio format has its own set of filters. The
name of the audio format for which the filter should be applied can
be used as the prefix for the filter option. Example:

        para_audiod -f 'mp3:prebuffer --duration 300'

The "mp3" prefix above is actually interpreted as a POSIX extended
regular expression. Therefore

        para_audiod -f '.:prebuffer --duration 300'

activates the prebuffer filter for all supported audio formats (because
"." matches all audio formats) while

        para_audiod -f 'wma|ogg:prebuffer --duration 300'

activates it only for wma and ogg streams.

Decoders
~~~~~~~~

For each supported audio format there is a corresponding filter
which decodes audio data in this format to 16 bit PCM data which
can be directly sent to the sound device or any other software that
operates on undecoded PCM data (visualizers, equalizers etc.). Such
filters are called _decoders_ in general, and xxxdec is the name of
the paraslash decoder for the audio format xxx. For example, the mp3
decoder filter is called mp3dec.

Note that the output of the decoder is about 10 times larger than
its input. This means that filters that operate on the decoded audio
stream have to deal with much more data than filters that transform
the audio stream before it is fed to the decoder.

Paraslash relies on external libraries for most decoders, so these
libraries must be installed for the decoder to be included in the
para_filter and para_audiod executables. The oggdec filter depends
on the libogg and libvorbis libraries for example.

Forward error correction
~~~~~~~~~~~~~~~~~~~~~~~~

As already mentioned REFERENCE(Streaming protocols, earlier),
paraslash uses forward error correction (FEC) for the unreliable UDP
and DCCP transports. FEC is a technique which was invented already
in 1960 by Reed and Solomon and which is widely used for the parity
calculations of storage devices (RAID arrays). It is based on the
algebraic concept of finite fields, today called Galois fields, in
honour of the mathematician Galois (1811-1832). The FEC implementation
of paraslash is based on code by Luigi Rizzo.

Although the details require a sound knowledge of the underlying
mathematics, the basic idea is not hard to understand: For positive
integers k and n with k < n it is possible to compute for any k given
data bytes d_1, ..., d_k the corresponding r := n -k parity bytes p_1,
..., p_r such that all data bytes can be reconstructed from *any*
k bytes of the set

        {d_1, ..., d_k, p_1, ..., p_r}.

FEC-encoding for unreliable network transports boils down to slicing
the audio stream into groups of k suitably sized pieces called _slices_
and computing the r corresponding parity slices. This step is performed
in para_server which then sends both the data and the parity slices
over the unreliable network connection. If the client was able
to receive at least k of the n = k + r slices, it can reconstruct
(FEC-decode) the original audio stream.

From these observations it is clear that there are three different
FEC parameters: The slice size, the number of data slices k, and the
total number of slices n. It is crucial to choose the slice size
such that no fragmentation of network packets takes place because
FEC only guards against losses and reordering but fails if slices are
received partially.

FEC decoding in paralash is performed through the fecdec filter which
usually is the first filter (there can be other filters before fecdec
if these do not alter the audio stream).


Volume adjustment (amp and compress)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

The amp and the compress filter both adjust the volume of the audio
stream. These filters operate on uncompressed audio samples. Hence
they are usually placed directly after the decoding filter. Each
sample is multiplied with a scaling factor (>= 1) which makes amp
and compress quite expensive in terms of computing power.

*amp*

The amp filter amplifies the audio stream by a fixed scaling factor
that must be known in advance. For para_audiod this factor is derived
from the amplification field of the audio file's entry in the audio
file table while para_filter uses the value given at the command line.

The optimal scaling factor F for an audio file is the largest real
number F >= 1 such that after multiplication with F all samples still
fit into the sample interval [-32768, 32767]. One can use para_filter
in combination with the sox utility to compute F:

        para_filter -f mp3dec -f wav < file.mp3 | sox -t wav - -e stat -v

The amplification value V which is stored in the audio file table,
however, is an integer between 0 and 255 which is connected to F
through the formula

        V = (F - 1) * 64.

To store V in the audio file table, the command

        para_client -- touch -a=V file.mp3

is used. The reader is encouraged to write a script that performs
these computations :)

*compress*

Unlike the amplification filter, the compress filter adjusts the volume
of the audio stream dynamically without prior knowledge about the peak
value. It maintains the maximal volume of the last n samples of the
audio stream and computes a suitable amplification factor based on that
value and the various configuration options. It tries to chose this
factor such that the adjusted volume meets the desired target level.

Note that it makes sense to combine amp and compress.

Misc filters (wav and prebuffer)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

These filters are rather simple and do not modify the audio stream at
all. The wav filter is only useful with para_filter and in connection
with a decoder. It asks the decoder for the number of channels and the
sample rate of the stream and adds a Microsoft wave header containing
this information at the beginning. This allows to write wav files
rather than raw PCM files (which do not contain any information about
the number of channels and the sample rate).

The prebuffer filter simply delays the output until the given time has
passed (starting from the time the first byte was available in its
input queue) or until the given amount of data has accumulated. It
is mainly useful for para_audiod if the standard parameters result
in buffer underruns.

Both filters require almost no additional computing time, even when
operating on uncompressed audio streams, since data buffers are simply
"pushed down" rather than copied.

Examples
~~~~~~~~

-> Decode an mp3 file to wav format:

        para_filter -f mp3dec -f wav < file.mp3 > file.wav

-> Amplify a raw audio file by a factor of 1.5:

        para_filter -f amp --amp 32 < foo.raw > bar.raw

------
Output
------

Once an audio stream has been received and decoded to PCM format,
it can be sent to a sound device for playback. This part is performed
by paraslash _writers_ which are described in this chapter.

Writers
~~~~~~~

A paraslash writer acts as a data sink that consumes but does not
produce audio data. Paraslash writers operate on the client side and
are contained in para_audiod and in the stand-alone tool para_write.

The para_write program reads uncompressed audio data from STDIN. If
this data starts with a wav header, sample rate, sample format and
channel count are read from the header. Otherwise CD audio (44.1KHz
16 bit little endian, stereo) is assumed but this can be overridden
by command line options. para_audiod, on the other hand, obtains
the sample rate and the number of channels from the decoder.

Like receivers and filters, each writer has an individual set of
command line options, and for para_audiod writers can be configured
per audio format separately. It is possible to activate more than
one writer for the same stream simultaneously.

OS-dependent APIs
~~~~~~~~~~~~~~~~~

Unfortunately, the various flavours of Unix on which paraslash
runs on have different APIs for opening a sound device and starting
playback. Hence for each such API there is a paraslash writer that
can play the audio stream via this API.

*ALSA*. The _Advanced Linux Sound Architecture_ is only available on
Linux systems. Although there are several mid-layer APIs in use by
the various Linux distributions (ESD, Jack, PulseAudio), paraslash
currently supports only the low-level ALSA API which is not supposed
to be change. ALSA is very feature-rich, in particular it supports
software mixing via its DMIX plugin. ALSA is the default writer on
Linux systems.

*OSS*. The _Open Sound System_ is the only API on *BSD Unixes and
is also available on Linux systems, usually provided by ALSA as an
emulation for backwards compatibility. This API is rather simple but
also limited. For example only one application can open the device
at any time. The OSS writer is activated by default on BSD Systems.

*OSX*. Mac OS X has yet another API called CoreAudio. The OSX writer
for this API is only compiled in on such systems and is of course
the default there.

*FILE*. The file writer allows to capture the audio stream and
write the PCM data to a file on the file system rather than playing
it through a sound device. It is supported on all platforms and is
always compiled in.

*AO*. _Libao_ is a cross-platform audio library which supports a wide
variety of platforms including PulseAudio (gnome), ESD (Enlightened
Sound Daemon), AIX, Solaris and IRIX.  The ao writer plays audio
through an output plugin of libao.

Examples
~~~~~~~~

-> Use the OSS writer to play a wav file:

        para_write --writer oss < file.wav

-> Enable ALSA software mixing for mp3 streams

        para_audiod --writer 'mp3:alsa -d plug:swmix'


---
Gui
---

para_gui executes an arbitrary command which is supposed to print
status information to STDOUT. It then displays this information in
a curses window. By default the command

        para_audioc -- stat -p

is executed, but this can be customized via the --stat-cmd option. In
particular it possible to use

        para_client -- stat -p

to make para_gui work on systems on which para_audiod is not running.

Key bindings
~~~~~~~~~~~~

It is possible to bind keys to arbitrary commands via custom
key-bindings. Besides the internal keys which can not be changed (help,
quit, loglevel, version...), the following flavours of key-bindings
are supported:

        - 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>".

The general form of a key binding is

        key_map k:m:c

which maps key k to command c using mode m. Mode may be x, d or p
for external, display and paraslash commands, respectively.

Themes
~~~~~~

Currently there are only two themes for para_gui. It is easy, however,
to add more themes. To create a new theme one has to define the
position, color and geometry for for each status item that should be
shown by this theme. See gui_theme.c for examples.

The "." and "," keys are used to switch between themes.

Examples
~~~~~~~~

-> Show server info:

        key_map "i:p:si"

-> Jump to the middle of the current audio file by pressing F5:

        key_map "<F5>:p:jmp 50"

-> vi-like bindings for jumping around:

        key_map "l:p:ff 10"
        key_map "h:p:ff 10-"
        key_map "w:p:ff 60"
        key_map "b:p:ff 60-"

-> Print the current date and time:

        key_map "D:d:date"

-> Call other curses programs:

        key_map "U:x:aumix"
        key_map "!:x:/bin/bash"
        key_map "^E:x:/bin/sh -c 'vi ~/.paraslash/gui.conf'"

-----------
Development
-----------

Tools
~~~~~

In order to compile the sources from the git repository (rather than
from tar balls) and for contributing non-trivial changes to the
paraslash project, some additional tools should be installed on a
developer machine.

http://git.or.cz/ (git). As described in more detail REFERENCE(Git
branches, below), the git source code management tool is used for
paraslash development. It is necessary for cloning the git repository
and for getting updates.

ftp://ftp.gnu.org/pub/gnu/m4/ (m4). Some input files for gengetopt
are generated from templates by the m4 macro processor.

ftp://ftp.gnu.org/pub/gnu/autoconf/ (autoconf) GNU autoconf creates
the configure file which is shipped in the tarballs but has to be
generated when compiling from git.

http://www.triptico.com/software/grutatxt.html (grutatxt). The
HTML version of this manual and some of the paraslash web pages are
generated by the grutatxt plain text to HTML converter. If changes
are made to these text files the grutatxt package must be installed
to regenerate the HTML files.

http://www.stack.nl/~dimitri/doxygen/ (doxygen). The documentation
of paraslash's C sources uses the doxygen documentation system. The
conventions for documenting the source code is described in the
REFERENCE(Doxygen, Doxygen section).

ftp://ftp.gnu.org/pub/gnu/global (global). This is used to generate
browsable HTML from the C sources. It is needed by doxygen.

Git branches
~~~~~~~~~~~~

Paraslash has been developed using the git source code management
tool since 2006. Development is organized roughly in the same spirit
as the git development itself, as described below.

The following text passage is based on "A note from the maintainer",
written by Junio C Hamano, the maintainer of git.

There are four branches in the paraslash repository that track the
source tree: "master", "maint", "next", and "pu".

The "master" branch is meant to contain what is well tested and
ready to be used in a production setting. There could occasionally be
minor breakages or brown paper bag bugs but they are not expected to
be anything major, and more importantly quickly and easily fixable.
Every now and then, a "feature release" is cut from the tip of this
branch, named with three dotted decimal digits, like 0.4.2.

Whenever changes are about to be included that will eventually lead to
a new major release (e.g. 0.5.0), a "maint" branch is forked off from
"master" at that point. Obvious, safe and urgent fixes after the major
release are applied to this branch and maintenance releases are cut
from it. New features never go to this branch. This branch is also
merged into "master" to propagate the fixes forward.

A trivial and safe enhancement goes directly on top of "master".
New development does not usually happen on "master", however.
Instead, a separate topic branch is forked from the tip of "master",
and it first is tested in isolation; Usually there are a handful such
topic branches that are running ahead of "master". The tip of these
branches is not published in the public repository to keep the number
of branches that downstream developers need to worry about low.

The quality of topic branches varies widely. Some of them start out as
"good idea but obviously is broken in some areas" and then with some
more work become "more or less done and can now be tested by wider
audience". Luckily, most of them start out in the latter, better shape.

The "next" branch is to merge and test topic branches in the latter
category.  In general, this branch always contains the tip of "master".
It might not be quite rock-solid production ready, but is expected to
work more or less without major breakage. The maintainer usually uses
the "next" version of paraslash for his own pleasure, so it cannot
be _that_ broken. The "next" branch is where new and exciting things
take place.

The two branches "master" and "maint" are never rewound, and "next"
usually will not be either (this automatically means the topics that
have been merged into "next" are usually not rebased, and you can find
the tip of topic branches you are interested in from the output of
"git log next"). You should be able to safely build on top of them.

However, at times "next" will be rebuilt from the tip of "master" to
get rid of merge commits that will never be in "master". The commit
that replaces "next" will usually have the identical tree, but it
will have different ancestry from the tip of "master".

The "pu" (proposed updates) branch bundles the remainder of the
topic branches.  The "pu" branch, and topic branches that are only in
"pu", are subject to rebasing in general.  By the above definition
of how "next" works, you can tell that this branch will contain quite
experimental and obviously broken stuff.

When a topic that was in "pu" proves to be in testable shape, it
graduates to "next".  This is done with

        git checkout next
        git merge that-topic-branch

Sometimes, an idea that looked promising turns out to be not so good
and the topic can be dropped from "pu" in such a case.

A topic that is in "next" is expected to be polished to perfection
before it is merged to "master".  Similar to the above, this is
done with

        git checkout master
        git merge that-topic-branch
        git branch -d that-topic-branch

Note that being in "next" is not a guarantee to appear in the next
release (being in "master" is such a guarantee, unless it is later
found seriously broken and reverted), nor even in any future release.

Coding Style
~~~~~~~~~~~~

The preferred coding style for paraslash coincides more or less
with the style of the Linux kernel. So rather than repeating what is
written XREFERENCE(http://www.kernel.org/doc/Documentation/CodingStyle,
there), here are the most important points.

        - Burn the GNU coding standards.
        - Never use spaces for indentation.
        - Tabs are 8 characters, and thus indentations are also 8 characters.
        - Don't put multiple assignments on a single line.
        - Avoid tricky expressions.
        - Don't leave whitespace at the end of lines.
        - The limit on the length of lines is 80 columns.
        - Use K&R style for placing braces and spaces:

                if (x is true) {
                        we do y
                }

        - Use a space after (most) keywords.
        - Do not add spaces around (inside) parenthesized expressions.
        - Use one space around (on each side of) most binary and ternary operators.
        - Do not use cute names like ThisVariableIsATemporaryCounter, call it tmp.
        - Mixed-case names are frowned upon.
        - Descriptive names for global variables are a must.
        - Avoid typedefs.
        - Functions should be short and sweet, and do just one thing.
        - The number of local variables shouldn't exceed 10.
        - Gotos are fine if they improve readability and reduce nesting.
        - Don't use C99-style "// ..." comments.
        - Names of macros defining constants and labels in enums are capitalized.
        - Enums are preferred when defining several related constants.
        - Always use the paraslash wrappers for allocating memory.
        - If the name of a function is an action or an imperative.
          command, the function should return an error-code integer
          (<0 means error, >=0 means success). If the name is a
          predicate, the function should return a "succeeded" boolean.


Doxygen
~~~~~~~

Doxygen is a documentation system for various programming
languages. The paraslash project uses Doxygen for generating the API
reference on the web pages, but good source code documentation is
also beneficial to people trying to understand the code structure
and the interactions between the various source files.

It is more illustrative to look at the source code for examples than
to describe the conventions for documenting the source in this manual,
so we only describe which parts of the code need doxygen comments,
but leave out details on documentation conventions.

As a rule, only the public part of the C source is documented with
Doxygen. This includes structures, defines and enumerations in header
files as well as public (non-static) C functions.  These should be
documented completely. For example each parameter and the return
value of a public function should get a descriptive comment.

No doxygen comments are necessary for static functions and for
structures and enumerations in C files (which are used only within
this file). This does not mean, however, that those entities need
no documentation at all. Instead, common sense should be applied to
document what is not obvious from reading the code.

--------
Appendix
--------

Network protocols
~~~~~~~~~~~~~~~~~

*IP*. The _Internet Protocol_ is the primary networking protocol
used for the Internet. All protocols described below use IP as the
underlying layer. Both the prevalent IPv4 and the next-generation
IPv6 variant are being deployed actively worldwide.

*Connection-oriented and connectionless protocols*. Connectionless
protocols differ from connection-oriented ones in that state
associated with the sending/receiving endpoints is treated
implicitly. Connectionless protocols maintain no internal knowledge
about the state of the connection. Hence they are not capable of
reacting to state changes, such as sudden loss or congestion on the
connection medium. Connection-oriented protocols, in contrast, make
this knowledge explicit. The connection is established only after
a bidirectional handshake which requires both endpoints to agree
on the state of the connection, and may also involve negotiating
specific parameters for the particular connection. Maintaining an
up-to-date internal state of the connection also in general means
that the sending endpoints perform congestion control, adapting to
qualitative changes of the connection medium.

*Reliability*. In IP networking, packets can be lost, duplicated,
or delivered out of order, and different network protocols handle
these problems in different ways. We call a transport-layer protocol
_reliable_, if it turns the unreliable IP delivery into an ordered,
duplicate- and loss-free delivery of packets. Sequence numbers
are used to discard duplicates and re-arrange packets delivered
out-of-order. Retransmission is used to guarantee loss-free
delivery. Unreliable protocols, in contrast, do not guarantee ordering
or data integrity.

*Classification*. With these definitions the protocols which are used
by paraslash for steaming audio data may be classified as follows.

        - HTTP/TCP: connection-oriented, reliable,
        - UDP: connectionless, unreliable,
        - DCCP: connection-oriented, unreliable.

Below we give a short descriptions of these protocols.

*TCP*. The _Transmission Control Protocol_ provides reliable,
ordered delivery of a stream and a classic window-based congestion
control. In contrast to UDP and DCCP (see below), TCP does not have
record-oriented or datagram-based syntax, i.e. it provides a stream
which is unaware and independent of any record (packet) boundaries.
TCP is used extensively by many application layers. Besides HTTP (the
Hypertext Transfer Protocol), also FTP (the File Transfer protocol),
SMTP (Simple Mail Transfer Protocol), SSH (Secure Shell) all sit on
top of TCP.

*UDP*. The _User Datagram Protocol_ is the simplest transport-layer
protocol, built as a thin layer directly on top of IP. For this reason,
it offers the same best-effort service as IP itself, i.e. there is no
detection of duplicate or reordered packets. Being a connectionless
protocol, only minimal internal state about the connection is
maintained, which means that there is no protection against packet
loss or network congestion. Error checking and correction (if at all)
are performed in the application.

*DCCP*. The _Datagram Congestion Control Protocol_ combines the
connection-oriented state maintenance known from TCP with the
unreliable, datagram-based transport of UDP. This means that it
is capable of reacting to changes in the connection by performing
congestion control, offering multiple alternative approaches. But it
is bound to datagram boundaries (the maximum packet size supported
by a medium), and like UDP it lacks retransmission to protect
against loss. Due to the use of sequence numbers, it is however
able to react to loss (interpreted as a congestion indication) and
to ignore out-of-order and duplicate packets. Unlike TCP it allows
to negotiate specific, binding features for a connection, such as
the choice of congestion control: classic, window-based congestion
control known from TCP is available as CCID-2, rate-based, "smooth"
congestion control is offered as CCID-3.

*HTTP*. _The Hypertext Transfer Protocol_ is an application layer
protocol on top of TCP. It is spoken by web servers and is most often
used for web services.  However, as can be seen by the many Internet
radio stations and YouTube/Flash videos, http is by far not limited to
the delivery of web pages only. Being a simple request/response based
protocol, the semantics of the protocol also allow the delivery of
multimedia content, such as audio over http.

*Multicast*. IP multicast is not really a protocol but a technique
for one-to-many communication over an IP network. The challenge is to
deliver information to a group of destinations simultaneously using
the most efficient strategy to send the messages over each link of
the network only once. This has benefits for streaming multimedia:
the standard one-to-one unicast offered by TCP/DCCP means that
n clients listening to the same stream also consume n-times the
resources, whereas multicast requires to send the stream just once,
irrespective of the number of receivers.  Since it would be costly to
maintain state for each listening receiver, multicast often implies
connectionless transport, which is the reason that it is currently
only available via UDP.

License
~~~~~~~

Paraslash is licensed under the GPL, version 2. Most of the code
base has been written from scratch, and those parts are GPL V2
throughout. Notable exceptions are FEC and the WMA decoder. See the
corresponding source files for licencing details for these parts. Some
code sniplets of several other third party software packages have
been incorporated into the paraslash sources, for example log message
coloring was taken from the git sources. These third party software
packages are all published under the GPL or some other license
compatible to the GPL.

Acknowledgements
~~~~~~~~~~~~~~~~

Many thanks to Gerrit Renker who read an early draft of this manual
and contributed significant improvements.

----------
References
----------

Articles
~~~~~~~~
        - Reed, Irving S.; Solomon, Gustave (1960),
        XREFERENCE(http://kom.aau.dk/~heb/kurser/NOTER/KOFA01.PDF,
        Polynomial Codes over Certain Finite Fields), Journal of the
        Society for Industrial and Applied Mathematics (SIAM) 8 (2):
        300-304, doi:10.1137/0108018)

RFCs
~~~~

        - XREFERENCE(http://www.ietf.org/rfc/rfc768.txt, RFC 768) (1980):
        User Datagram Protocol
        - XREFERENCE(http://www.ietf.org/rfc/rfc791.txt, RFC 791) (1981):
        Internet Protocol
        - XREFERENCE(http://www.ietf.org/rfc/rfc2437.txt, RFC 2437) (1998):
        RSA Cryptography Specifications
        - XREFERENCE(http://www.ietf.org/rfc/rfc4340.txt, RFC 4340)
        (2006): Datagram Congestion Control Protocol (DCCP)
        - XREFERENCE(http://www.ietf.org/rfc/rfc4341.txt, RFC 4341) (2006):
        Congestion Control ID 2: TCP-like Congestion Control
        - XREFERENCE(http://www.ietf.org/rfc/rfc4342.txt, RFC 4342) (2006):
        Congestion Control ID 3: TCP-Friendly Rate Control (TFRC)
        - XREFERENCE(http://www.ietf.org/rfc/rfc6716.txt, RFC 6716) (2012):
        Definition of the Opus Audio Codec

Application web pages
~~~~~~~~~~~~~~~~~~~~~

        - XREFERENCE(http://paraslash.systemlinux.org/, paraslash)
        - XREFERENCE(http://xmms2.org/wiki/Main_Page, xmms)
        - XREFERENCE(http://www.mpg123.de/, mpg123)
        - XREFERENCE(http://gstreamer.freedesktop.org/, gstreamer)
        - XREFERENCE(http://www.icecast.org/, icecast)
        - XREFERENCE(http://beesbuzz.biz/code/audiocompress.php, Audio Compress)

External documentation
~~~~~~~~~~~~~~~~~~~~~~

        - XREFERENCE(http://kernel.org/pub/linux/kernel/people/hpa/raid6.pdf,
        H. Peter Anvin: The mathematics of Raid6)
        - XREFERENCE(http://info.iet.unipi.it/~luigi/fec_ccr.ps.gz,
        Luigi Rizzo: Effective Erasure Codes for reliable Computer
        Communication Protocols)

Code
~~~~
        - XREFERENCE(http://info.iet.unipi.it/~luigi/vdm.tar.gz,
        Original FEC implementation by Luigi Rizzo)