ecasound - sample editor, multitrack recorder, fx-processor, etc.

ecasound [ general_options ] { [ chain_setup ] [ effect_setup ] [ input_setup ] [ output_setup ] }

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.

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, that defaults to 271. See ECA_LOGGER class documentation for more detailed info about various debug_level values.

-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).

--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. 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

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. 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, 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

Works in the same way as the -i option. If no outputs are specified, the default output device is used (see ~/.ecasoundrc). 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

-erc:from-channel, to-channel

Copy channel 'from_channel' to 'to_channel'. If 'to_channel' doesn't exist, it is created. Channel indexing is started from 1.

-erm:to-channel

Mix all channels to channel 'to_channel'. If 'to_channel' doesn't exist, it is created. Channel indexing is started from 1.

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. You really should see examples.html for some more info.

-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).

ECASOUND If defined, some utility programs and scripts will use the ECASOUND environment as the default path to ecasound executable.

~/.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.

ecatools(1), ecasound-iam(1) ecasoundrc(5), "HTML docs in the Documentation subdirectory"

See file BUGS. If ecasound behaves weirdly, try to increase the debug level to see what's going on.

Kai Vehmanen, <kvehmanen -at- eca -dot- cx>