Ubuntu Manpage: ecasound - sample editor, multitrack recorder, fx-processor, etc.

Provided by: ecasound_2.4.6.1-1_i386

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 Global options, can be used
in ecasound chainsetup files (.ecs).

GLOBAL OPTIONS

-c Starts ecasound in interactive mode. In interactive mode you can
control ecasound with simple commands ("start", "stop", "pause",
etc.). See ecasound-iam(1).

-C Disables ecasound’s interactive mode (see ’-c’ and ’-K’).

-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.

-D Print all debug information to stderr (unbuffered, plain output
without ncurses).

-q Quiet mode, no output. Same as -d:0.

-s[:]chainsetup-file
Create a new chainsetup from file ’chainsetup-file’ and add it
to the current session. Chainsetup can contain inputs, outputs,
chains, effects, controllers, etc. A session, on the other hand,
contains all the chainsetups. Although only one chainsetup can
be connected at a time, you can switch between them on-the-fly.

--daemon
Enable ecasound’s daemon mode. If this option is given, ecasound
will allow clients to connect to the running ecasound session
using a TCP/IP network connection. The clients can both observe
and control the session.

Warning! As there is no access control implemented, be sure to
block ecasound’s port in your firewall if the machine running
ecasound is connected to a public network! Otherwise anyone can
connect to your ecasound sessions.

--daemon-port
Set the TCP port used by the daemon mode. By default ecasound
will use port number 2868.

--nodaemon
Disable ecasound’s daemon mode. This is the default.

--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.

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(5).

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 size of buffer in samples (must be an exponent of 2).
This is quite an important option. For real-time processing, you
should 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), values between 512 - 4096 often give
better results. Default is 1024.

-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 default sampling parameters. These are used for all
following input and output files or until another -f is
specified. If no -f option is present, ecasound will use the
default audio parameters (see ecasoundrc(5) man page). Notice
that when opening existing audio objects, either for input or
output, the default audio parameters set with -f are ignored if
objects provide sufficient header information (as is the case
for wav, aiff, etc formats). For output objects, the -x option
can be used to completely overwrite existing files (in other
words, with -x, default audio parameters set with -f are used).

Sample format is given as a 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 endianess 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".

The 4th parameter ’interleaving’ should either be ’i’ (default)
for interleaved stream format, or ’n’ for noninterleaved.

-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. Connecting multiple inputs to the same chain isn’t
possible. 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 files (.mp2,.mp3). Also,
formats supported by the SGI audiofile library: AIFF (.aiff,
.aifc, .aif) and Sun/NeXT audio files (.au, .snd). 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, you can’t output to module formats supported by MikMod
(this should be obvious).

OBJECT TYPE SPECIFIC NOTES

ALSA devices
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
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
If enabled at compile-time, ecasound supports audio input and
output using aRts audio server. Option syntax is -i:arts,
-o:arts.

Ecasound Wave Files - .ewf
A simple wrapper class for handling other audio objects. See
ecasound user’s guide for more detailed information.

JACK input/outputs
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.

Ecasound provides multiple ways to communicate with JACK
servers. To directly input or output to soundcard, use -i
jack_alsa and -o jack_alsa. To communicate with other apps, use
jack_auto,remote_clientname. To just create ports without making
any automatic connections, there are jack and
jack_generic,local_portprefix.

Additionally global JACK options can be set using
-G:jack,client_name,operation_mode. ’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" (the
default) which is a combination of the two previous modes.

More details about ecasound’s JACK support can be found from
ecasound user’s guide.

Libaudiofile
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
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 devices make it possible to route data between chains.
Option syntax is -[io][:]loop,id_number. If you add a loop
output with id ’1’, all data written to this output is routed to
all loop inputs with id ’1’. You can attach the same loop device
to multiple inputs and outputs.

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
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 - access object of different sample rates
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.

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 - process audio data backwards
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.

System standard streams and named pipes
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.

Typeselect - overriding object type settings
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.

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(1) 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 and stop to MIDI device-id ’device_id’.

-Mss Sends MIDI-sync to the selected MIDI-device. Note! Ecasound will
not send MIDI-clock, but only 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(5) 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 and -eas effects. Also prints a
statistics table containing info about stereo-image and how
different sample values are used.

-evp Peak amplitude watcher. Maintains peak information for each
processed channels. Peak information is resetted 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-%
Amplifies signal by amplify-% percent.

-eac:amplify-%,channel
Amplifies signal of channel ’channel’ by amplify-% percent.
’channel’ ranges from 1...n where n is the total number of
channels.

-eaw:amplify-%,max-clipped-samples
Amplifies signal by amplify-% percent. If number of consecutive
clipped samples (resulting sample has the largest amplitude
possible) reaches ’max-clipped-samples’, a warning will be
issued.

-eal:limit-%
Limiter effect. Limits audio level to ’limit-%’.

-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. Basicly 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.

-chmix:to-channel
Mix all 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.

dir(-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-%’ determines how much
effected (wet) signal is mixed to the original. ’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). Plugins are located in shared library (.so)
files in /usr/local/share/ladspa (configured in ecasoundrc(5)).
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 expect plugin’s unique id-number is used. It is
guaranteed that these id-numbers are unique among all LADSPA
plugins.

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 trucated to zero
length.

-ge:open-threshold-%, close-thold-%,volume-mode
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. If ’value_mode’ is ’rms’, average RMS volume is used.
Otherwise peak average is used. When closed, passing audio
buffers are trucated to zero length.

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,freq,mode,point-pairs,start-value,end-
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.
’start-value’ and ’end-value’ are used as border values. 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(5)).

-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,start-value,end-
value,point_count,pos1,value1,...,posN,valueN
Generic linear envelope. This controller source can be used to
map custom envelopes to chain operator parameters. All ’posX’
parameters are given as seconds (from start of the stream).
’valueX’ parameters must be in the range [0,1].

-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(5)). 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).

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).

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.

FILES

              ~/.ecasound The default directory  for  ecasound  user  resource
              files.  See the ecasoundrc(5) 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://eca.cx/ecasound/Documentation/examples.html.

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>>

                                  06.12.2006                       ecasound(1)

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