Provided by: ecasound_2.9.3-4build2_amd64 
NAME
ecasound - sample editor, multitrack recorder, fx-processor, etc.
SYNOPSIS
ecasound [ general_options ] { [ chain_setup ] [ effect_setup ] [ input_setup ] [ output_setup ] }
DESCRIPTION
Ecasound is a software package designed for multitrack audio processing. It can be used for simple tasks
like audio playback, recording and format conversions, as well as for multitrack effect processing,
mixing, recording and signal recycling. Ecasound supports a wide range of audio inputs, outputs and
effect algorithms. Effects and audio objects can be combined in various ways, and their parameters can
be controlled by operator objects like oscillators and MIDI-CCs. A versatile console mode user-interface
is included in the package.
OPTIONS
Note! All options except those mentioned in ecasound options and Global options, can be used in ecasound
chainsetup files (.ecs).
ECASOUND OPTIONS
These options are parsed and handled by the ecasound frontend binary and are not passed to backend
library. This means that these options may not work in other applications that use ecasound
libraries for their functionality.
-c Starts ecasound in interactive mode. In interactive mode you can control ecasound with simple
commands ("start", "stop", "pause", etc.). See ecasound-iam .
-C Disables ecasound’s interactive mode (see ’-c’ and ’-K’).
-D Print all debug information to stderr (unbuffered, plain output without ncurses).
-s[:]chainsetup-file
Create a new chainsetup from file ’chainsetup-file’ and add it to the current session. Chainsetup
files commonly have a filename ending to the ’.ecs’ extension. A chainsetup can contain inputs,
outputs, chains, effects, controllers -- i.e. objects one one specific configuration of audio
processing elements. A session, on the other hand, is a collection of one or more chainsetups.
Only one of the chainsetups may be connected (i.e. it can be run/processed). But it is possible to
have another chainsetup select (i.e. can be configured) while other one is current connteced (i.e.
running).
-E "cmd1 [[args] ; cmd2 args ; ... ; cmdN]"
Execute a set of Ecasound Interactive mode (EIAM) commands at launch. These commands are executed
immediately after ecasound is started. If the command line contains sufficient options to create a
valid chainsetup that will be executed, the launch commands are executed after the other command
line options are parsed, but before the processing engine is started. Note that this command is a
feature of the ecasound frontend binary and not supported by the library backend. This means that
other clients may not support the ’-E’ option, and also that the launch commands are not saved as
part of chainsetup or session state.
--server
Enables the so called NetECI mode, in which ecasound can be controlled remotely over a socket
connection. When activated, clients can connect to the running ecasound session, and use
interactive mode commands to control and observe ecasound processing.
The NetECI protocol is defined in Ecasound’s Programmer Guide
One example client using this feature is ecamonitor(1). This utility is included in the Ecasound
distribution package (requires a working Python environment).
Warning! If the machine running ecasound, is connected to a public network, be sure to block
ecasound’s port in your firewall! As there is no access control implemented for incoming
connections, anyone can otherwise connect, control and observe your ecasound sessions. This
option replaces ’--daemon’ (deprecated in 2.6.0).
--server-tcp-port=NNN
Set the TCP port used by the daemon mode. By default ecasound will use port number 2868. This
option replaces ’--daemon-port’ (deprecated in 2.6.0).
--no-server
Disable ecasound’s daemon mode. This is the default. This option replaces ’--nodaemon’
(deprecated in 2.6.0).
--osc-udp-port=NNN
Enables support for Open Source Control (OSC). Ecasound will listen for incoming OSC messages on
UDP port NNN. Ecasound’s OSC interface is documented at:
<http://ecasound.git.sourceforge.net/git/gitweb.cgi?p=ecasound/ecasound;a=blob;f=Documentation/ecasound_osc_interface.txt;hb=HEAD>
Note that OSC support is still experimental and the interface might change in later versions of
Ecasound.
This option was added to ecasound 2.7.0.
--keep-running,-K
Do not exit when processing is finished/stopped. Only affects non-interactive operating mode (see
-c/-C). Option added to ecasound 2.4.2.
--help,-h
Show this help.
--version
Print version info.
GLOBAL OPTIONS
-d, -dd, -ddd
Increase the amount of printed debug messages. -d adds some verbosity, while -ddd results in very
detailed output.
-d:debug_level
Set the debug level mask to ’debug_level’. This a bitmasked value with the following classes:
errors (1), info (2), subsystems (4), module_names (8), user_objects (16), system_objects 32,
functions (64), continuous (128) and eiam_return_values (256). Default is 271 (1+2+4+8+256). See
sourcode documentation for the ECA_LOGGER class for more detailed information.
-R[:]path-to-file
Use ecasound resource file (see ecasoundrc man page) ’path-to-file’ as the only source of setting
resource value. Specifying this option will disable the normal policy of querying both global and
user (if exists) resource files.
-q Quiet mode, no output. Same as -d:0.
GENERAL CHAINSETUP OPTIONS
-a:chainname1, chainname2, ...
Selects active signal chains. All inputs and outputs following this ’-a’ option are assigned to
selected chains (until a new -a option is specified). When adding effects, controllers and other
chain operators, only one chain can be selected at a time. If no -a option has been given, chain
’default’ is used instead when adding objects. Chain name ’all’ is also reserved. It will cause
all existing chains to be selected. By giving multiple -a options, you can control to which chains
effects, inputs and outputs are assigned to. Look at the EXAMPLES section for more detailed info
about the usage of this option.
-n:name
Sets the name of chainsetup to ’name’. If not specified, defaults either to "command-line-setup"
or to the file name from which chainsetup was loaded. Whitespaces are not allowed.
-x Truncate outputs. All output object are opened in overwrite mode. Any existing files will be
truncated.
-X Open outputs for updating. Ecasound opens all outputs - if target format allows it - in readwrite
mode.
-z:feature
Enables ’feature’. Most features can be disabled using notation -z:nofeature. ’-z:db,dbsize’
enables double-buffering for audio objects that support it (dbsize=0 for default, otherwise buffer
size in sample frames). ’-z:nodb’ disables double-buffering. ’-z:intbuf’ and ’-z:nointbuf’
control whether extra internal buffering is allowed for realtime devices. Disabling this can
reduce latency times in some situations. With ’-z:xruns’, processing will be halted if an
under/overrun occurs. ’-z:multitrack’ and ’z:nomultitrack’ can be used to force ecasound to enable
or disable multitrack-mode. In rare cases you may want to explicitly specify the recording offset
with ’-z:multitrack,offset-in-samples’. The offset is the amount of samples skipped when recording
from real-time inputs. ’-z:psr’ enables the precise-sample-rates mode for OSS-devices.
’-z:mixmode,sum’ enables mixing mode where channels are mixed by summing all channels. The default
is ’-z:mixmode,avg’, in which channels are mixed by averaging. Mixmode selection was first added
to ecasound 2.4.0. See ecasoundrc man page.
CHAINSETUP BUFFERING AND PERFORMANCE OPTIONS
-B:buffering_mode
Selects the default buffering mode. Mode is one of: ’auto’ (default), ’nonrt’, ’rt’,
’rtlowlatency’.
-b:buffer_size
Sets the processing engine buffer size in samples. The size must be an exponent of 2, and it is
independent of channel count (e.g. -b:1024 at 48kHz will result in 21.333ms buffer length whether
input is mono, stereo or 5.1).
This is an important option as this defines the length of one processing engine iteration and
affects ecasound behaviour in many ways. If not explicitly specified, ecasound will try to choose
an optimal value based on current buffering mode (see -B option). For real-time processing, you
can try to set this as low as possible to reduce the processing delay. Some machines can handle
buffer values as low as 64 and 128. In some circumstances (for instance when using oscillator
envelopes) small buffer sizes will make envelopes act more smoothly. When not processing in
real-time (all inputs and outputs are normal files), larger values may help to avoid buffer
overruns, lower CPU usage and/or otherwise improve performance.
Note that when any JACK input/outputs are used, the buffer size setting is overridden and set to
period/buffer size reported by JACK server (e.g. jackd’s ’-p’ option). It is not possible to turn
off this behaviour.
If not explicitly specified, the default buffer size is chosen based on current buffering mode
(see -B).
-r:sched_priority
Use realtime scheduling policy (SCHED_FIFO). This is impossible if ecasound doesn’t have root
priviledges. Beware! This gives better performance, but can cause total lock-ups if something goes
wrong. The ’sched_priority’ can be omitted (0=omitted). If given, this is the static priority to
the highest priority ecasound thread. Other ecasound threads run with priority
’sched_priority-1...n’. Value ’-1’ can be used to disable raised-priority mode.
-z:feature
Relevant features are -z:db,xxx (-z:nodb) and -z:intbuf (-z:nointbuf). See section General
chainsetup options for details.
PROCESSING CONTROL
-t:seconds
Sets processing time in seconds (doesn’t have to be an integer value). If processing time isn’t
set, engine stops when all inputs are finished. This option is equivalent to the ’cs-set-length’
EIAM command. A special-case value of ’-1’ will set the chainsetup length according to the longest
input object.
-tl Enables looping. When processing is finished, engine will start again from beginning. This option
is equivalent to the ’cs-loop’ EIAM command.
INPUT/OUTPUT SETUP
See ecasound user’s guide for more detailed documentation.
-G:mgrtype,optstring
Sets options for audio object manager type ’mgrtype’. For available options, see "OBJECT TYPE
SPECIFIC NOTES" below.
-f:sample_format,channel,sample-rate,interleaving
Sets the audio stream parameters for subsequent audio objects. To set different parameters for
different audio objects, multiple ’-f’ options have to be specified (note the ordering, the ’-f’
options should precede the audio objects for them to have any effect). See documentation for ’-i’
and ’-o’ options.
When an audio object is opened (e.g. a file or sound device is opened, or connection is made to a
sound server), the audio stream parameters are passed to the object. It should be noted that not
all audio objects allow one to set any or all of the parameters. For instance when opening
existing audio files, many file formats have a header describing the file audio parameters. In
these cases the audio file header overrides the parameters passed with ’-f’ option. Similarly when
creating JACK inputs and outputs, the JACK server mandates the sampling rate and sample format.
If no ’-f’ option is specified, or some of the argument fields are left empty (e.g.
’-f:,2,44100’), ecasound will use default values. These default values are defined in ecasoundrc
configuration file. See ecasoundrc(5) manual page.
Note that ecasound opens out files by default in update mode. Unless option ’-x’ (overwrite
outputs) option is given, audio parameters of an existing audio file take preference over the
params set with ’-f’.
Sample format is given as a formatted string. The first letter is either "u", "s" and "f"
(unsigned, signed, floating point). The following number specifies sample size in bits. If sample
is little endian, "_le" is added to the end. Similarly if big endian, "_be" is added. If
endianness is not specified, host byte-order is used. Currently supported formats are "u8" (same
as "8"), "s16_le" (same as "16"), "s16_be", "s24_le", "s24_be", "s32_le", "s32_be", "f32_le" and
"f32_be". An empty string "" picks the system default sample format.
The 4th parameter defines the channel layout. The available options are ’i’ (interleaved’ and ’n’
(noninterleaved). With the noninterleaved setting, ecasound will process samples one channel at a
time, and the blocksize is set with ’-b’. The default setting is ’i’.
-y:seconds
Sets starting position for last specified input/output. If you need more flexible control over
audio objects, you should use the .ewf format.
-i[:]input-file-or-device[,params]
Specifies a new input source that is connected to all selected chains (chains are selected with
’-a:...’). Connecting multiple inputs to the same chain is not possible, but one input can be
connected to multiple chains. Input can be a a file, device or some other audio object (see
below). If the input is a file, its type is determined using the file name extension. If the
object name contains any commas, the name must be enclosed in backquotes to avoid confusing the
parser. Currently supported formats are RIFF WAVE files (.wav), audio-cd tracks (.cdr), ecasound
EWF files (.ewf), RAW audio data (.raw) and MPEG audio files (.mp2,.mp3). More audio formats are
supported via libaudiofile and libsndfile libraries (see documentation below). MikMod is also
supported (.xm, .mod, .s3m, .it, etc). MIDI files (.mid) are supported using Timidity++.
Similarly Ogg Vorbis (.ogg) can be read, and written if ogg123 and vorbize tools are installed;
FLAC files (.flac) with flac command-line tools or using libsndfile; and AAC files
(.aac/.m4a/.mp4) with faad2/faac tools. Supported realtime devices are OSS audio devices
(/dev/dsp*), ALSA audio and loopback devices and JACK audio subsystem. If no inputs are specified,
the first non-option (doesn’t start with ’-’) command line argument is considered to be an input.
-o[:]output-file-or-device[,params]
Works in the same way as the -i option. If no outputs are specified, the default output device is
used (see ~/.ecasoundrc). If the object name contains any commas, the name must be enclosed in
backquotes to avoid confusing the parser. Note, many object types do not support output (e.g.
MikMod, MIDI and many others).
OBJECT TYPE SPECIFIC NOTES
ALSA devices - ’alsa’
When using ALSA drivers, instead of a device filename, you need to use the following option
syntax: -i[:]alsa,pcm_device_name.
ALSA direct-hw and plugin access - ’alsahw’, ’alsaplugin’
It’s also possible to use a specific card and device combination using the following notation:
-i[:]alsahw,card_number,device_number,subdevice_number. Another option is the ALSA PCM plugin
layer. It works just like the normal ALSA pcm-devices, but with automatic channel count and sample
format conversions. Option syntax is -i[:]alsaplugin,card_number,device_number,subdevice_number.
aRts input/output - ’arts’
If enabled at compile-time, ecasound supports audio input and output using aRts audio server.
Option syntax is -i:arts, -o:arts.
Audio file sequencing - ’audioloop’, ’select’, ’playat’
Ecasound provides a set of special audio object types that can be used for temporal sequencing of
audio files - i.e. looping, playing only a select portion of a file, playing file at a specific
time, and other such operation.
Looping is possible with -i:audioloop,file.ext,params. The file name (or any object type
understood by Ecasound) given as the second parameter is played back continuously looping back to
the beginning when the end of file is reached. Any additional parameters given are passed
unaltered to the file object. Parameters 3...N are passed as is to the child object (i.e. "-i
audioloop,foo.wav,bar1,bar2" will pass parameters "bar1,bar2" to the "foo.wav" object.
To select and use only a specific segment of an audio object, the
-i:select,start-time,duration,file.ext,params can be used. This will play "duration" of
"file.ext", starting at "start-time". The time values should be given as seconds (e.g. "2.25", or
as samples (e.g. "25000sa"). Parameters 4...N are passed as is to the child object.
To play an audio object at a given moment in time, the -i:playat,play-at-time,file.ext,params can
be used. This will play "file.ext" after position reaches "play-at-time". The time values should
be given as seconds (e.g. "2.25", or as samples (e.g. "25000sa"). Parameters 2...N are passed as
is to the child object.
Ecasound Wave Files (EWF) - ’*.ewf’
A special file format that allows one to slice and loop full (or segments) of audio files. This
format is specific to Ecasound. See ecasound user’s guide for more detailed information.
See also audio object types ’audioloop’, ’select’ and ’playat’.
JACK input/outputs - Overview
JACK is a low-latency audio server that can be used to connect multiple independent audio
application to each other. It is different from other audio server efforts in that it has been
designed from the ground up to be suitable for low-latency professional audio work.
JACK input/outputs - ’jack’
Ecasound provides multiple ways to communicate with JACK servers. To create a JACK input or output
object, one should use -i jack and -o jack. These create JACK client ports "ecasound:in_N" and
"ecasound:out_n" respectively (’N’ is replaced by the channel number). Ecasound automatically
creates one JACK port for each channel (number of channels is set with -f:bits,channels,rate
option).
It is important to note that by default JACK ports are not connected anywhere (e.g. to soundcard
input/outputs, or to other apps). One thus has to connect the ports with an external program (e.g.
"QJackCtl" or "jack_connect").
JACK input/outputs - ’jack,clientname,portprefix’
"jack,clientname" For simple use scanerios, ecasound provides a way to autoconnect the ecasound
ports. This can be done with by giving the peer client name as the second parameter to the "jack"
object, e.g. -o jack,clientname. As an example, -o jack,system will create an output that is
automatically connected to outputs of the default system soundcard. The client parameter can be
omitted, in which case no automatic connections are made.
If one needs to change the port prefix (e.g. "in" in client name "ecasound:in_N"), the prefix can
be specified as the third parameter to "jack" object, e.g. -o jack,,fxout. Also the third
parameter can be omitted, in which case the default prefixes "in" and "out" are used.
JACK input/outputs - ’jack_multi’
A variant of ’jack’ object type is ’jack_multi’. The full object syntax is
jack_multi,destport1,...,destportN. When a ’jack_multi’ object is connected to a JACK server,
first channel of the object is connected to JACK port ’destport1’, second to ’destport2’ and so
forth. For instance "-f:32,2,44100 -o jack_multi,foo:in,bar:in" creates a stereo ecasound output
object, with its left and right channels routed to two difference JACK clients. The destination
ports must be active when the ecasound engine is launched, or otherwise the connections cannot be
established. If destination ports are not specified for all channels, or zero length strings are
given, those ports are not connected at launch by ecasound.
JACK input/outputs - ’jack_alsa’, ’jack_auto’, ’jack_generic’ (**deprecated since 2.6.0**)
Ecasound 2.5 and older supported "jack_alsa", "jack_auto" and "jack_generic" object types, but
these are now replaced by a more generic "jack" interface, and thus are now deprecated (they work
but are no longer documented).
JACK input/outputs - client options
Additionally global JACK options can be set using -G:jack,client_name,operation_mode option.
’client_name’ is the name used when registering ecasound to the JACK system. If ’operation_mode’
is "notransport", ecasound will ignore any transport state changes in the JACK-system; in mode
"send" it will send all start, stop and position-change events to other JACK clients; in mode
"recv" ecasound will follow JACK start, stop and position-change events; and mode "sendrecv" which
is a combination of the two previous modes.
If not explicitly set, in interactive mode (’-c’ option), the default mode is "sendrecv", while in
batchmode default is "notransport". In both cases the mode can be changed with -G option as
described above.
More details about ecasound’s JACK support can be found from Ecasound User’s Guide.
Libaudiofile - ’audiofile’
If libaudiofile support was enabled at compile-time, this option allows you to force Ecasound to
use libaudiofile for reading/writing a certain audio file. Option syntax is
-i:audiofile,foobar.ext (same for -o).
Libsndfile - ’sndfile’
If libsndfile support was enabled at compile-time, this option allows you to force Ecasound to use
libsndfile for reading/writing a certain audio file. Option syntax is
-i:sndfile,foobar.ext[,.format-ext] (same for -o). The optional third parameter "format" can be
used to override the audio format (for example you can create an AIFF file with filename
"foo.wav").
Loop device - ’loop’
Loop devices make it possible to route (loop back) data between chains. Option syntax is
-[io][:]loop,tag. If you add a loop output with tag ’1’, all data written to this output is routed
to any loop input with tag ’1’. The tag can be either numerical (e.g. ’-i:loop,1’) or a string
(e.g. "-i:loop,vocals"). Like with other input/output objects, you can attach the same loop device
to multiple chains and this way split/mix the signal.
Note: this ’loop’ device is different from ’audioloop’ (latter added to ecasound v2.5.0).
Mikmod - ’mikmod’
If mikmod support was enabled at compile-time, this option allows you to force Ecasound to use
Mikmod for reading/writing a certain module file. Option syntax is -i:mikmod,foobar.ext.
Null inputs/outputs - ’null’
If you specify "null" or "/dev/null" as the input or output, a null audio device is created. This
is useful if you just want to analyze sample data without writing it to a file. There’s also a
realtime variant, "rtnull", which behaves just like "null" objects, except all i/o is done at
realtime speed.
Resample - ’resample’
Object type ’resample’ can be used to resample audio object’s audio data to match the sampling
rate used in the active chainsetup. For example, ecasound -f:16,2,44100 -i resample,22050,foo.wav
-o /dev/dsp, will resample file from 22.05kHz to 44.1kHz and write the result to the soundcard
device. Child sampling rate can be replaced with keyword ’auto’. In this case ecasound will try to
query the child object for its sampling rate. This works with files formats such as .wav which
store meta information about the audio file format. To use ’auto’ in the previous example,
ecasound -f:16,2,44100 -i resample,auto,foo.wav -o /dev/dsp.
Parameters 4...N are passed as is to the child object (i.e. "-i resample,22050,foo.wav,bar1,bar2"
will pass parameters "bar1,bar2" to the "foo.wav" object.
If ecasound was compiled with support for libsamplerate, you can use ’resample-hq’ to use the
highest quality resampling algorithm available. To force ecasound to use the internal resampler,
’resampler-lq’ (low-quality) can be used.
Reverse - ’reverse’
Object type ’reverse’ can be used to reverse audio data coming from an audio object. As an
example, ecasound -i reverse,foo.wav -o /dev/dsp will play ’foo.wav’ backwards. Reversing output
objects is not supported. Note! Trying to reverse audio object types with really slow seek
operation (like mp3), works extremely badly. Try converting to an uncompressed format (wav or
raw) first, and then do reversation.
Parameters 3...N are passed as is to the child object (i.e. "-i reverse,foo.wav,bar1,bar2" will
pass parameters "bar1,bar2" to the "foo.wav" object.
System standard streams and named pipes - ’stdin’, ’stdout’
You can use standard streams (stdin and stdout) by giving stdin or stdout as the file name. Audio
data is assumed to be in raw/headerless (.raw) format. If you want to use named pipes, create them
with the proper file name extension before use.
Tone generator - ’tone’
To generate a test tone, input -i:tone,type,freq,duration-secs can be used. Parameter ’type’
specifies the tone type: currently only ’sine’ is supported. The ’freq’ parameter sets the
frequency of the generated tone and ’duration-secs’ the length of the generated stream. Specifying
zero, or a negative value, as the duration will produce an infinite stream. This feature was first
added to Ecasound 2.4.7.
Typeselect - ’typeselect’
The special ’typeselect’ object type can be used to override how ecasound maps filename extensions
and object types. For instance ecasound -i typeselect,.mp3,an_mp3_file.wav -o /dev/dsp. would
play the file ’an_mp3_file.wav’ as an mp3-file and not as an wav-file as would happen without
typeselect.
Parameters 4...N are passed as is to the child object (i.e. "-i typeselect,.au,foo.wav,bar1,bar2"
will pass parameters "bar1,bar2" to the "foo.wav" object.
MIDI SETUP
MIDI I/O devices - general
If no MIDI-device is specified, the default MIDI-device is used (see ecasoundrc(5)).
-Md:rawmidi,device_name
Add a rawmidi MIDI I/O device to the setup. ’device_name’ can be anything that can be accessed
using the normal UNIX file operations and produces raw MIDI bytes. Valid devices are for example
OSS rawmidi devices (/dev/midi00), ALSA rawmidi devices (/dev/snd/midiC2D0), named pipes (see
mkfifo man page), and normal files.
-Md:alsaseq,sequencer-port
Adds a ALSA MIDI sequencer port to the setup. ’sequencer-port’ identifies a port to connect to. It
can be numerical (e.g. 128:1), or a client name (e.g. "KMidimon").
-Mms:device_id
Sends MMC start ("Deferred Play") and stop ("Stop") with device ID ’device_id’.
While Ecasound does not directly support syncing transport state to incoming MMC messages, this
can be achieved by connecting Ecasound to JACK input/outputs, and using a tool such as JackMMC and
JackCtlMMC ( see <http://jackctlmmc.sourceforge.net/>) to convert MMC messages into JACK transport
change events.
-Mss Sends MIDI-sync (i.e. "MIDI Start" and "MIDI Stop" system realtime messages) .to the selected
MIDI-device. Notice that as Ecasound will not send MIDI-clock, but only the start and stop
messages.
EFFECT SETUP
PRESETS
Ecasound has a powerful effect preset system that allows you create new effects by combining basic
effects and controllers. See ecasound user’s guide for more detailed information.
-pf:preset_file.eep
Uses the first preset found from file ’preset_file.eep’ as a chain operator.
-pn:preset_name
Find preset ’preset_name’ from global preset database and use it as a chain operator. See
ecasoundrc man page for info about the preset database.
SIGNAL ANALYSIS
-ev Analyzes sample data to find out how much the signal can be amplified without clipping. The
resulting percent value can be used as a parameter to ’-ea’ (amplify). A statistical summary,
containing info about the stereo-image and distribution of sample values, is printed out at the
end of processing.
-evp Peak amplitude watcher. Maintains peak information for each processed channels. Peak information
is reset on every read.
-ezf Finds the optimal value for DC-adjusting. You can use the result as a parameter to -ezx effect.
GENERAL SIGNAL PROCESSING ALGORITHMS
-eS:stamp-id
Audio stamp. Takes a snapshot of passing audio data and stores it using id ’stamp-id’ (integer
number). This data can later be used by controllers and other operators.
-ea:amplify%
Adjusts the signal amplitude to ’amplify%’ percent (linear scale, i.e. individual samples are
multiplied by ’amplify%/100’). See also ’-eadb’.
-eac:amplify%,channel
Amplifies signal of channel ’channel’ by amplify-% percent (linear scale, i.e. individual samples
are multiplied by ’amplify%/100’). ’channel’ ranges from 1...n where n is the total number of
channels. See also ’-eadb’.
-eadb:gain-dB[,channel]
Adjusts signal level by ’gain-dB’, with a gain of 0dB having no effect to the signal, negative
gains attenuating the signal and positive gain values amplifying it. The ’channel’ parameter
(1...n) is optional. If ’channel’ parameter is specified, and its value is nonzero, gain is only
applied to the given channel (1...n).
-eaw:amplify%,max-clipped-samples
Amplifies signal by amplify-% percent (linear scale, i.e. individual samples are multiplied by
’amplify%/100’). If number of consecutive clipped samples (resulting sample value is outside the
nominal [-1,1] range), a warning will be issued.
-eal:limit-%
Limiter effect. Limits audio level to ’limit-%’ (linear scale) with values equal or greater than
100% resulting in no change to the signal.
-ec:rate,threshold-%
Compressor (a simple one). ’rate’ is the compression rate in decibels (’rate’ dB change in input
signal causes 1dB change in output). ’threshold’ varies between 0.0 (silence) and 1.0 (max
amplitude).
-eca:peak-level-%, release-time-sec, fast-crate, crate
A more advanced compressor (original algorithm by John S. Dyson). If you give a value of 0 to any
parameter, the default is used. ’peak-level-%’ essentially specifies how hard the peak limiter is
pushed. The default of 69% is good. ’release_time’ is given in seconds. This compressor is very
sophisticated, and actually the release time is complex. This is one of the dominant release time
controls, but the actual release time is dependent on a lot of factors regarding the dynamics of
the audio in. ’fastrate’ is the compression ratio for the fast compressor. This is not really the
compression ratio. Value of 1.0 is infinity to one, while the default 0.50 is 2:1. Another
really good value is special cased in the code: 0.25 is somewhat less than 2:1, and sounds super
smooth. ’rate’ is the compression ratio for the entire compressor chain. The default is 1.0, and
holds the volume very constant without many nasty side effects. However the dynamics in music are
severely restricted, and a value of 0.5 might keep the music more intact.
-enm:threshold-level-%,pre-hold-time-msec,attack-time-msec,post-hold-time-msec,release-time-msec
Noise gate. Supports multichannel processing (each channel processed separately). When signal
amplitude falls below ’threshold_level_%’ percent (100% means maximum amplitude), gate is
activated. If the signal stays below the threshold for ’th_time’ ms, it’s faded out during the
attack phase of ’attack’ ms. If the signal raises above the ’threshold_level’ and stays there over
’hold’ ms the gate is released during ’release’ ms.
-ei:pitch-shift-%
Pitch shifter. Modifies audio pitch by altering its length.
-epp:right-%
Stereo panner. Changes the relative balance between the first two channels. When ’right-%’ is 0,
only signal on the left (1st) channel is passed through. Similarly if it is ’100’, only right
(2nd) channel is let through.
-ezx:channel-count,delta-ch1,...,delta-chN
Adjusts the signal DC by ’delta-chX’, where X is the channel number. Use -ezf to find the optimal
delta values.
ENVELOPE MODULATION
-eemb:bpm,on-time-%
Pulse gate (pulse frequency given as beats-per-minute).
-eemp:freq-Hz,on-time-%
Pulse gate.
-eemt:bpm,depth-%
Tremolo effect (tremolo speed given as beats-per-minute).
FILTER EFFECTS
-ef1:center_freq, width
Resonant bandpass filter. ’center_freq’ is the center frequency. Width is specified in Hz.
-ef3:cutoff_freq, reso, gain
Resonant lowpass filter. ’cutoffr_freq’ is the filter cutoff frequency. ’reso’ means resonance.
Usually the best values for resonance are between 1.0 and 2.0, but you can use even bigger values.
’gain’ is the overall gain-factor. It’s a simple multiplier (1.0 is the normal level). With high
resonance values it often is useful to reduce the gain value.
-ef4:cutoff, resonance
Resonant lowpass filter (3rd-order, 36dB, original algorithm by Stefan M. Fendt). Simulates an
analog active RC-lowpass design. Cutoff is a value between [0,1], while resonance is between
[0,infinity).
-efa:delay-samples,feedback-%
Allpass filter. Passes all frequencies with no change in amplitude. However, at the same time it
imposes a frequency-dependent phase-shift.
-efc:delay-samples,radius
Comb filter. Allows the spikes of the comb to pass through. Value of ’radius’ should be between
[0, 1.0).
-efb:center-freq,width
Bandpass filter. ’center_freq’ is the center frequency. Width is specified in Hz.
-efh:cutoff-freq
Highpass filter. Only frequencies above ’cutoff_freq’ are passed through.
-efi:delay-samples,radius
Inverse comb filter. Filters out the spikes of the comb. There are ’delay_in_samples-2’ spikes.
Value of ’radius’ should be between [0, 1.0). The closer it is to the maximum value, the deeper
the dips of the comb are.
-efl:cutoff-freq
Lowpass filter. Only frequencies below ’cutoff_freq’ are passed through.
-efr:center-freq,width
Bandreject filter. ’center_freq’ is the center frequency. Width is specified in Hz.
-efs:center-freq,width
Resonator. ’center_freq’ is the center frequency. Width is specified in Hz. Basically just another
resonating bandpass filter.
CHANNEL MIXING / ROUTING
-chcopy:from-channel, to-channel
Copy channel ’from_channel’ to ’to_channel’. If ’to_channel’ doesn’t exist, it is created. Channel
indexing starts from 1. Option added to ecasound 2.4.5.
-chmove:from-channel, to-channel
Copy channel ’from_channel’ to ’to_channel’, and mutes the source channel ’from_channel’. Channel
indexing starts from 1. Option added to ecasound 2.4.5.
-chorder:ch1,...,chN
Reorder, omit and/r duplicate chain channels. The resulting audio stream has total of ’N’
channels. Each parameter specifies the source channel to use for given output channel. As an
example, ’-chorder:2,1’ would reverse the channels of a stereo stream (’out1,out2’ = ’in2,in1’).
Specifying the same source channel multiple times is allowed. For example, ’-chorder:2,2’ would
route the second channel to both two output channels (’out1,out2’ = ’in2,in2’). If ’chX’ is zero,
the given channel ’X’ will be muted in the output stream. Option added to ecasound 2.7.0.
-chmix:to-channel
Mix all source channels to channel ’to_channel’. If ’to_channel’ doesn’t exist, it is created.
Channel indexing starts from 1. Option added to ecasound 2.4.5.
-chmute:channel
Mutes the channel ’channel’. Channel indexing starts from 1. Option added to ecasound 2.4.5.
-erc:from-channel,to-channel
Deprecated, see -chcopy.
-erm:to-channel
Deprecated, see -chmix.
TIME-BASED EFFECTS
-etc:delay-time-msec,variance-time-samples,feedback-%,lfo-freq
Chorus.
-etd:delay-time-msec,surround-mode,number-of-delays,mix-%,feedback-%
Delay effect. ’delay time’ is the delay time in milliseconds. ’surround-mode’ is a integer with
following meanings: 0 = normal, 1 = surround, 2 = stereo-spread. ’number_of_delays’ should be
obvious. Beware that large number of delays and huge delay times need a lot of CPU power. ’mix-%’
expresses the mix balance between the original and delayed signal, with 0 meaning no delayed
signal, 100 meaning no original signal, and 50 (the default) achieving an equal balance.
’feedback-%’ represents how much of the signal is recycled in each delay or, if you prefer, at
what rate the repeated snippet of delayed audio fades. Note that sufficiently low feedback values
may result in a number of audible repetitions lesser than what you have specified for
’number_of_delays’, especially if you have set a low value for ’mix-%’. By default the value for
this parameter is 100% (No signal loss.).
-ete:room_size,feedback-%,wet-%
A more advanced reverb effect (original algorithm by Stefan M. Fendt). ’room_size’ is given in
meters, ’feedback-%’ is the feedback level given in percents and ’wet-%’ is the amount of reverbed
signal added to the original signal.
-etf:delay-time-msec
Fake-stereo effect. The input signal is summed to mono. The original signal goes to the left
channels while a delayed version (with delay of ’delay time’ milliseconds) is goes to the right.
With a delay time of 1-40 milliseconds this adds a stereo-feel to mono-signals.
-etl:delay-time-msec,variance-time-samples,feedback-%,lfo-freq
Flanger.
-etm:delay-time-msec,number-of-delays,mix-%
Multitap delay. ’delay time’ is the delay time in milliseconds. ’number_of_delays’ should be
obvious. ’mix-%’ determines how much effected (wet) signal is mixed to the original.
-etp:delay-time-msec,variance-time-samples,feedback-%,lfo-freq
Phaser.
-etr:delay-time,surround-mode,feedback-%
Reverb effect. ’delay time’ is the delay time in milliseconds. If ’surround-mode’ is ’surround’,
reverbed signal moves around the stereo image. ’feedback-%’ determines how much effected (wet)
signal is fed back to the reverb.
LADSPA-PLUGINS
-el:plugin_unique_name,param-1,...,param-N
Ecasound supports LADSPA-effect plugins (Linux Audio Developer’s Simple Plugin API). Parameters
1..N are set as values of the plugin’s control ports.
If plugin has more than one audio input and/or output port, only one plugin is instance is
created, and the chain channels are fed to the same plugin instance. If plugin has at most one
input and at most one output audio port, a separate plugin instance is created for each channel of
the ecasound chain (e.g. for a stereo audio channel, two LADSPA plugins of same type are created,
with one per channel).
Plugins are located in shared library (.so) files. Ecasound looks for plugins in
@prefix@/lib/ladspa (e.g. "/usr/local/lib/ladspa"), directories listed in environment variable
LADSPA_PATH. Plugin search path can be configured also via ecasoundrc, see ecasoundrc(5) man page.
One shared library file can contain multiple plugin objects, but every plugin has a unique plugin
name. This name is used for selecting plugins.
See LAD mailing list web site for more info about LADSPA. Other useful sites are LADSPA home page
and LADSPA documentation.
-eli:plugin_unique_number,param-1,...,param-N
Same as above (-el) expect plugin’s unique id-number is used. It is guaranteed that these
id-numbers are unique among all LADSPA plugins.
LV2 PLUGINS
-elv2:plugin-id-uri,param-1,...,param-N
Ecasound also supports LV2 audio plugins. LV2 plugins are identified by a globally unique,
case-sensitive identifier.
If plugin has more than one audio input and/or output port, only one plugin is instance is
created, and the chain channels are fed to the same plugin instance. If plugin has at most one
input and at most one output audio port, a separate plugin instance is created for each channel of
the ecasound chain (e.g. for a stereo audio channel, two LV2 plugins of same type are created,
with one per channel).
LV2 is a plugin standard for audio systems.
GATE SETUP
-gc:start-time,len
Time crop gate. Initially gate is closed. After ’start-time’ seconds has elapsed, gate opens and
remains open for ’len’ seconds. When closed, passing audio buffers are truncated to zero length.
-ge:open-threshold-%,close-thold-%,volume-mode,reopen-count
Threshold gate. Initially gate is closed. It is opened when volume goes over ’othreshold’ percent.
After this, if volume drops below ’cthold’ percent, gate is closed and won’t be opened again,
unless the ’reopen-count’ is set to anything other than zero. If ’value_mode’ is ’rms’, average
RMS volume is used. Otherwise peak average is used. When closed, passing audio buffers are
truncated to zero length. If the ’reopen-count’ is set to a positive number, then the gate will
restart its operation that many times. So for example, a reopen count of 1 will cause up to 2
openings of the gate. A negative value for ’reopen-count’ will result in the gate reopening
indefinitely. The ’reopen-count’ is invaluable in recording vinyl and tapes, where you can set
things up and then recording starts whenever the needle is on the vinyl, and stops when it’s off.
As many sides as you like can be recorded in one session. You will need to experiment with buffer
lengths and start/stop levels to get reliable settings for your equipment.
-gm:state
Manual gate. If ’state’ is 1, gate is open and all samples are passed through. If ’state’ is zero,
gate is closed an no samples are let through. This chain operator is useful when writing to an
output needs to be stopped dynamically (without stopping the whole engine).
CONTROL ENVELOPE SETUP
Controllers can be used to dynamically change effect parameters during processing. All controllers
are attached to the selected (=usually the last specified effect/controller) effect. The first
three parameters are common for all controllers. ’fx_param’ specifies the parameter to be
controlled. Value ’1’ means the first parameter, ’2’ the second and so on. ’start_value’ and
’end_value’ set the value range. For examples, look at the the EXAMPLES section.
-kos:fx-param,start-value,end-value,freq,i-phase
Sine oscillator with frequency of ’freq’ Hz and initial phase of ’i_phase’ times pi.
-kog:fx-param,start-value,end-value,freq,mode,point-pairs,first-value,last-value,pos1,value1,...
Generic oscillator. Frequency ’freq’ Hz, mode either ’0’ for static values or ’1’ for linear
interpolation. ’point-pairs’ specifies the number of ’posN’ - ’valueN’ pairs to include.
’first-value’ and ’last-value’ are used as border values (values for position 0.0/first and
position 1.0/last). All ’posN’ and ’valueN’ must be between 0.0 and 1.0. Also, for all ’posN’
values ’pos1 < pos2 < ... < posN’ must be true.
-kf:fx-param,start-value,end-value,freq,mode,genosc-number
Generic oscillator. ’genosc_number’ is the number of the oscillator preset to be loaded. Mode is
either ’0’ for static values or ’1’ for linear interpolation. The location for the preset file is
taken from ./ecasoundrc (see ecasoundrc man page).
-kl:fx-param,start-value,end-value,time-seconds
Linear envelope that starts from ’start_value’ and linearly changes to ’end_value’ during
’time_in_seconds’. Can be used for fadeins and fadeouts.
-kl2:fx-param,start-value,end-value,1st-stage-length-sec,2nd-stage-length-sec
Two-stage linear envelope, a more versatile tool for doing fade-ins and fade-outs. Stays at
’start_value’ for ’1st_stage_length’ seconds and then linearly changes towards ’end_value’ during
’2nd_stage_length’ seconds.
-klg:fx-param,low-value,high-value,point_count,pos1,value1,...,posN,valueN
Generic linear envelope. This controller source can be used to map custom envelopes to chain
operator parameters. Number of envelope points is specified in ’point_count’. Each envelope point
consists of a position and a matching value. Number of pairs must match ’point_count’ (i.e.
’N==point_count’). The ’posX’ parameters are given as seconds (from start of the stream). The
envelope points are specified as float values in range ’[0,1]’. Before envelope values are mapped
to operator parameters, they are mapped to the target range of ’[low-value,high-value]’. E.g. a
value of ’0’ will set operator parameter to ’low-value’ and a value of ’1’ will set it to
’high-value’. For the initial segment ’[0,pos1]’, the envelope will output value of ’value1’ (e.g.
’low-value’).
-km:fx-param,start-value,end-value,controller,channel
MIDI continuous controller (control change messages). Messages on the MIDI-channel ’channel’ that
are coming from controller number ’controller’ are used as the controller source. As recommended
by the MIDI-specification, channel numbering goes from 1 to 16. Possible controller numbers are
values from 0 to 127. The MIDI-device where bytes are read from can be specified using -Md option.
Otherwise the default MIDI-device is used as specified in ~ecasound/ecasoundrc (see ecasoundrc man
page). Defaults to /dev/midi.
-ksv:fx-param,start-value,end-value,stamp-id,rms-toggle
Volume analyze controller. Analyzes the audio stored in stamp ’stamp-id’ (see ’-eS:id’ docs), and
creates control data based on the results. If ’rms-toggle’ is non-zero, RMS-volume is used to
calculate the control value. Otherwise average peak-amplitude is used.
-kx This is a special switch that can be used when you need to control controller parameters with
another controller. When you specify -kx, the last specified controller will be set as the
control target. Then you just add another controller as usual.
INTERACTIVE MODE
See ecasound-iam(1) man page.
ENVIRONMENT
ECASOUND
If defined, some utility programs and scripts will use the ECASOUND environment as the default
path to ecasound executable.
ECASOUND_LOGFILE
Output all debugging messages to a separate log file. If defined, ECASOUND_LOGFILE defines the
logfile path. This is a good tool for debugging ECI/EIAM scripts and applications.
ECASOUND_LOGLEVEL
Select which messages are written to the logfile defined by ECASOUND_LOGFILE. The syntax for
-d:level is used. If not defined, all messages are written. Defaults to -d:319 (everything else
but ’functions (64)’ and ’continuous (128)’ class messages).
COLUMNS
Ecasound honors the COLUMNS environment variable when formatting printed trace messages. If
COLUMNS is not set, a default of 74 is used.
TMPDIR Some functions of Ecasound (e.g. "cs-edit" interactive command) require creation of temporary
files. By default, these files are created under "/tmp", but this can be overridden by setting the
TMPDIR environment variable.
RETURN VALUES
In interactive mode, ecasound always returns zero.
In non-interactive (batch) mode, a non-zero value is returned for the following errors:
1 Unable to create a valid chainsetup with the given parameters. Can be caused by invalid option
syntax, etc.
2 Unable to start processing. This can be caused by insufficient file permissions, inability to
access some system resources, etc.
3 Error during processing. Possible causes: output object has run out of free disk space, etc.
4 Error during process termination and/or cleanup. See section on ’SIGNALS’ for further details.
SIGNALS
When ecasound receives any of the POSIX signals SIGINT (ctrl-c), SIGHUP, SIGTERM or SIGQUIT, normal
cleanup and exit procedure is initiated. Here normal exit means that e.g. file headers are updated before
closing, helper processes are terminated in normal way, and so forth.
If, while doing the cleanup described above, ecasound receives another signal (of the same set of POSIX
signals), ecasound will skip the normal cleanup procedure, and terminate immediately. Any remaining
cleanup tasks will be skipped. Depending on the runtime state and configuration, this brute force exit
may have some side-effects. Ecasound will return exit code of ’4’ if normal cleanup was skipped.
Special case handling is applied to the SIGINT (ctrl-c) signal. If a SIGINT signal is received during
the cleanup procedure, ecasound will ignore the signal once, and emit a notice to ’stderr’ that cleanup
is already in progress. Any subsequent SIGINT signals will no longer get special handling, and instead
process will terminate immediately (and possibly without proper cleanup).
FILES
~/.ecasound The default directory for ecasound user resource files. See the ecasoundrc (5) man page man
page.
*.ecs Ecasound Chainsetup files. Syntax is more or less the same as with command-line arguments.
*.ecp Ecasound Chain Preset files. Used for storing effect and chain operator presets. See ecasound
user’s guide for more better documentation.
*.ews Ecasound Wave Stats. These files are used to cache waveform data.
EXAMPLES
Examples of how to perform common tasks with ecasound can be found at
http://nosignal.fi/ecasound/Documentation/examples.html.
SEE ALSO
ecatools (1) man page, ecasound-iam (1) man page ecasoundrc (5) man page, "HTML docs in the Documentation
subdirectory"
BUGS
See file BUGS. If ecasound behaves weirdly, try to increase the debug level to see what’s going on.
AUTHOR
Kai Vehmanen, <kvehmanen -at- eca -dot- cx <kvehmanen -at- eca -dot- cx>>
05.05.2011 ecasound(1)