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mirrors: eca.cx (eu) / seul.org (us) / sourceforge.net (us)
Remember to also check out the Ecasound Tutorials and Articles [alt link] page, the ecasound(1) [alt link] manpage, and the Ecasound User’s Guide.
The console mode user-interface, ecasound, is used in all the following examples. Other ecasound frontends may use a different syntax, but the basic principles are the still the same as long as ecasound is used as the backend engine.
Version note: sections describing recently introduced features have a note about the minimum required version.
These first two commands do the exact same thing, conver somefile.wav to somefile.cdr (CDR is the CD-Audio track format). As no chains are specified, the default chain is created and used.
1. ecasound -i:somefile.wav -o:somefile.cdr 2. ecasound -i somefile.wav -o somefile.cdr
This is not a very useful example, but hopefully helps to understand the way chains work:
3. ecasound -a:1,2 -i somefile.wav -a:1 -o somefile.cdr -a:2 -o somefile.mp3
First, two new chains 1 and 2 (you can also use strings: '-a:some_name_with_no_whitespaces,some_other_name') are created. They are now the active chains. After this, input somefile.wav is connected to both these chains. The rest follows the same scheme. Chain '1' is set active and output somefile.cdr is attached to it. In the same way, somefile.mp3 is attached to chain '2'.
The last example is similar to the first two, but now ecasound is started in interactive mode:
4. ecasound -c -i somefile.wav -o somefile.cdr
Format Conversions - Resampling
1. ecasound -f:16,2,96000 -i resample,auto,foo44100.wav -o bar96k.wav 2. ecasound -f:16,2,44100 -i resample,auto,bar96k.wav -o foo44100.wav 3. ecasound -f:16,2,44100 -i resample-hq,48k,foo48k.wav -o bar.wav 4. ecasound -f:16,2,44100 -i resample,96k,third.raw -o foo44100.wav
To do resampling, a special 'resample' input type must be used. In the first example above, the 44100Hz file foo44100.wav is resampled to 96kHz and the result is written to bar96k.wav. In the second example the reverse is done, sample rate is converted from 96khz to 44100Hz. In the last example, the from rate has to be specified explicitly as raw audio files do not contain the necessary header information.
In the last example we do a 48000Hz to 44100Hz conversion using the 'resample-hq' input type. If Ecasound was built with support for the libsamplerate package, 'resample-hq' selects the high-quality conversion mode. In this mode the conversion process requires more CPU power, but the results are of higher quality. Using libsamplerate also improves quality of the default 'resample' mode.
Realtime Outputs (playing to a sound device)
Following are examples of recording from, and playing back to, ALSA sound device (Linux):
1. ecasound -i somefile.mp3 -o alsa 2. ecasound -i somefile.mp3 -o alsahw,0,0 3. ecasound -i somefile.mp3 -o alsaplugin,0,0 4. ecasound -i somefile.mp3 -o alsa,soundcard_name
ALSA sound subsystem provides multiple ways to select which audio device to use, and how it is accessed. A plain "alsa" will use the default ALSA sound device (depends on system configuration). Alternatively you can either specify "alsahw" (to indicate you want use the ALSA direct hardware interface), or "alsaplugin" (to utilize the ALSA plugin layer). Both accept card number and device number as parameters. Optionally also subdevice can be given. The plugin layer is able to handle some type conversions automatically. Yet another option is to specify the ALSA device name ('soundcard_name'). The device name must be defined in the ALSA configuration files (either in ~/.asoundrc or in the global settings file).
Following are examples of recording from, and playing back to, OSS sound device (Linux, BSDs and many other OS'es):
1. ecasound somefile.wav 2. ecasound -i somefile.wav 3. ecasound -i:somefile.wav 4. ecasound -i somefile.wav -o /dev/dsp
If you haven't touched your ~/.ecasound/ecasoundrc configuration file, these should all do the same thing, output somefile.wav to /dev/dsp using the default chain. If no inputs are specified, ecasound tries to use the first non-option argument on the command line as a default input. If no chains are specified, the chain 'default' is created and set active. If no outputs are specified, the default-output defined in ~/.ecasound/ecasoundrc is used. This is normally /dev/dsp.
See also section on JACK below.
Realtime Inputs (recording from a sound device)
1. ecasound -i /dev/dsp0 -o somefile.wav 2. ecasound -i alsa -o somefile.wav -c 3. ecasound -i alsahw,1,0 -o somefile.wav
These are simple examples of recording. When recording, it may be useful to run ecasound in interactive mode (-c).
See also section on JACK below.
Use with JACK audio server
1. ecasound -i foo.wav -o jack,system 2. ecasound -i foo.wav -o jack
This will create a separate JACK output port for each channel of foo.wav, and automatically connect these Ecasound ports to the system PCM output ports in the JACK server. The second example will create JACK output ports, but does not establish any connections (you have to do this yourself with jack_connect, qjackconnet, qjackctl, or other similar tool).
3. ecasound -c -i foo.wav -o jack,system -G:jack,eca_slave,recv
Like in previous examples, but the ecasound client name (as shown to other JACK clients) is set to "eca_slave", and ecasound is configured to react to incoming transport changes (play, stop, seek, etc). By default ecasound both sends and reacts to transport events.
Version note: the above describes the updated JACK interface introduced in Ecasound 2.6. The old interfaces, "jack_alsa", "jack_auto" and "jack_generic" are deprecated, but still available in current versions of Ecasound.
Ecasound is an extremely versatile tool when it comes to effect processing. After all, it was originally programmed for non-realtime signal processing. Here are some of examples:
1. ecasound -i somefile.mp3 -o alsa -ea:120 2. ecasound -a:default -i somefile.mp3 -o alsa -ea:120
These two perform the same thing: an mp3 input file is amplified to 120% (linear scale) and fed to ALSA default PCM sound device.
3.ecasound -i somefile.mp3 -o alsa -etr:40,0,55 -ea:120
Like the previous examples, but now a reverb effect, with a delay of 40 milliseconds, with surround mode disabled and mix ratio of 55%, is added to the chain before the amplify effect. In other words the signal is first processed with the reverb and then amplified. Any number of effects can be combined this way. Note that when a real-time input/output, like the ALSA PCM device in above example, is part of the setup, one must have enough CPU power to run the effect algorithms in real-time (at the speed of the sound device). When this happens, ecasound will start warning about buffer over- or underruns (often referred to as 'xruns'), and the audio output may be garbled. An easy way to work aroud processing power limitations is to split the processing into two steps: first apply the effects and write output to a file, and then as a separate step play the file to a sound device.
4. ecasound -a:1,2 -i somefile.mp3 -o alsa \ -a:1 -etr:40,0,55 -ea:120 \ -a:2 -efl:400
Ok, let's next do some parallel processing: two chains are created and the input and output files are connected to them. As a result, the input signal is processed with two sets of effects, and then mixed back together. You can create as many chains this way as you want.
Using controller sources with effects
1. ecasound -i somefile.wav -o alsa -ef3:800,1.5,0.9 -kos:1,400,4200,0.2,0 -kos:2,0.1,1.5,0.15,0 2. ecasound -i somefile.wav -o alsa -ef3:800,1.5,0.9 -km:1,400,4200,74,1 -km:2,0.1,1.5,71,1
The first example uses two sine oscillators ('-kos:parameter,range_low,range_high,speed_in_Hz,initial_phase') to control a resonant lowpass filter. The cutoff frequency varies betweeen 400 and 4200 Hz, while resonance varies between 0.1 and 1.5. The initial phase is 0 (times pi). The second example uses MIDI continuous controllers ('-km:parameter,range_low,range_high,controller_number,midi-channel') as controller sources. The ranges are the same as in the in first example. Controller numbers used are 74 (cutoff) and 71 (resonance). In other words you can use your synth's cutoff and resonance knobs.
It's also possible to control controllers with other controllers using the '-kx' option. Normally when you add a controller, you're controlling the last specified chain operator. '-kx' changes this. Let's take an example:
3. ecasound -i file.wav -o alsa -ea:100 -kos:1,0,100,0.5,0 -kx -kos:4,0.1,5,0.5,0
Same as before, but now another 0.5Hz sine oscillator is controlling the frequency of the first oscillator.
Now let's add a MIDI-controller (CC) to the mix. In the following, a sine oscillator is assigned to the cutoff frequency, while other controller is controlling the resonance. MIDI control change messages from controller 2 (range of 0-127) on channel 1 (range of 1-16 channels) are used to control the frequency of the sine oscillator (with MIDI control value range of 0-127 mapped to oscillator frequencies 0.5-1.5kHz).
4. ecasound -i file.wav -o alsa -ef3:1000,1.0,1.0 -kos:1,500,2000,1,0 \ -kos:2,0.2,1.0,0.5,0 \ -kx -km:1,0.1,1.5,2,1
1. ecasound -c -f:16,2,44100 \ -a:1 -i monitor-track.wav -o alsa \ -a:2 -i alsa -o new-track.wav
It really is this simple. Then a default sample format is set with '-f:bits,channels,sample_rate'. Now all that's left is to specify two chains: one for monitoring and one for recording. When using the above command, you need to have some way of monitoring the signal that is recorded. The preferred way is to utilize hw-monitoring of the audio device: unmute and adjust the line-in level with the mixer application (e.g. "alsamixer" or "alsamixergui"). If you want to use ecasound for digital monitoring, you have to add a separate chain for it:
2. ecasound -c -b:256 \ -a:1 -i monitor-track.wav \ -a:2,3 -i alsa \ -a:2 -o new-track.wav \ -a:1,3 -o alsa
You can always do test recordings until you find the optimal volume levels (using the soundcard mixer apps and adjusting source volume), but ecasound offers a better way to do this. This is a bit ugly, but what's most important, it works in text-mode:
3. ecasound -c -f:16,2,44100 -a:1 -i alsa -o alsa -ev
Basicly this just records from ALSA default PCM input, puts the signal through an analyze ('-ev') operator and outputs to ALSA output. The secret here is that you can get volume statistics with the estatus (or es) command in interactive mode. Newer ecasound versions (1.8.5 and newer) provide a separate 'ecasignalview' application, which can be used to monitor signal level in real-time.
Here's a few real-life mixdown examples.
1. ecasound -c \ -a:1 -i drums.wav \ -a:2 -i synth-background.wav \ -a:3 -i bass-guitar_take-2.ewf \ -a:4 -i brass-house-lead.wav \ -a:all -o alsa
First of all, interactive-mode is selected with '-c' to allow controlling of mixdown playback (starting, stopping, seeking and so forth). The mixdown consists of four inputs (all stereo). Each input is routed to a dedicated chains (named '1'...'4'). The chains are mixed by routing them all to a single output device (the ALSA default PCM device).
2. ecasound -c -r \ -a:1 -i drums.wav -ea:200 \ -a:2 -i synth-background.wav -epp:40 -ea:120 \ -a:3 -i bass-guitar_take-2.ewf -ea:75 \ -a:4 -i brass-house-lead.wav -epp:60 -ea:50 \ -a:1,2,3,4 -o loop,1 \ -a:5,6 -i loop,1 \ -a:5 -o alsa \ -a:6 -o current-mix.wav
This second example is more complex. The same inputs are used, but this time effects (amplify '-ea:mix_percent' and panning '-epp:left_right_balance') are also used. Additionally we want to route the full mix to both the ALSA sound device and to a file. As a single chain can be connected to at most one input and one output, use of a virtual loop device is needed in this example. The four input chains are first routed to a loop device 'loop,1' ('1' identifies the loop object instance). Then two new chains, '5' and '6', are defined. The loop device is set as input to both of these chains (using the '-a:5,6 -i loop,1' syntax). As the final step, the ALSA device is set as the output for chain '5' and the file 'current-mix.wav' as the output of chain '6'.
Cut, Copy and Paste
Here's a simple example where the first 60 seconds of bigfile.wav is written to part1.wav and the rest to part2.wav:
1.a) ecasound -i bigfile.wav -o part1.wav -t:60.0 b) ecasound -i bigfile.wav -y:60.0 -o part2.wav
If you want to combine these files back to one big file:
2. ecasound -i part2.wav -o part1.wav -y:500
In the above second example, part2.wav is appended to part1.wav.
Manipulating objects - looping, reversing, ...
To continuously loop an audio file, you can use 'audioloop':
1. ecasound -i audioloop,drumloop.wav -o alsa
To run the loop for 65.0secs, you can use '-t':
2. ecasound -i audioloop,drumloop.wav -o alsa -t:65.0
To play an audio file in reserve:
3. ecasound -i reverse,drumloop.wav -o alsa
The various operations can also be stacked. Let's first start by selecting a 5sec second out of a bigger file and play that out to the ALSA device.
4. ecasound -i select,5,10,audioclip.wav -o alsa
Next let's play that segment in reverse:
5. ecasound -i reverse,select,5,10,audioclip.wav -o alsa
But you can still keep adding new operators. Let's now loop the reversed segment:
Another available operator is "playat", which can be used to play an audio clip at a given moment in time (i.e. postpone the time it is played). A simple example of playing "audioclip.wav" at position 20sec (i.e. 20secs of silences and then file is played out from start).6. ecasound -i audioloop,reverse,select,5,10,audioclip.wav -o alsa
This operatoror can be combined with others as well. To select a 10sec clip, starting at 25sec (i.e. 25-35sec of "audioclip.wav"), and play it at 180.5sec (with total length of 180.5+10.0=190.5sec):7. ecasound -i playat,20.0,audioclip.wav -o alsa
8. ecasound -i playat,180.5,select,25,10.0,audioclip.wav -o alsa
Version note: 'audioloop', 'select' and 'playat' audio object types were added to ecasound version 2.5.0.
Ecasound chains can transport multiple channels of audio. The channel count of a chain is defined by the input and output connected to it. Normally effects and other chain operators operate on all channels of a chain. But it is also possible to operate on specific c hannels only. Here's an example of how to split a four channel input file into four separate one channel output files:
1. ecasound -a:1,2,3,4 -i 4-channel-file.wav \ -a:1 -f:16,1,44100 -o mono-1.wav \ -a:2 -f:16,1,44100 -o mono-2.wav -chcopy:2,1 \ -a:3 -f:16,1,44100 -o mono-3.wav -chcopy:3,1 \ -a:4 -f:16,1,44100 -o mono-4.wav -chcopy:4,1
Note that all four channels of '4-channel-file.raw' are routed to all the four chains '1' through '4'. At the output stage, all the output files are one channel (because of '-f:16,1,44100) and thus only the first channel of each chain is written to the output. To perform the correct routing, the '-chcopy' operator is used to copy source channel 'N' to first channel (which will be written to the output file). In chain '1' no channel copying is needed as the first channel already has the necessary contents.
Version note: '-chcopy' was introduced in version 2.4.5. In earlier versions '-erc' provides similar functionality.
To amplify a specific channel, '-eac' can be used. Here's an example where 3rd channel of a 4ch file is amplified by 150% (linear scale):
2. ecasound -i 4ch-infile.wav -ea:150,3 -o 4ch-outfile.wav
Signal Routing through External Devices
Signal routing is done using similar setups as is used for multirack recording. The only difference is that the realtime input(s) and output(s) are externally connected (fed through an external effect processor for instance).
1. ecasound -c -f:16,2,44100 \ -a:1 -i source-track.wav -o alsa \ -a:2 -i alsa -o processed-track.wav
LADSPA Effect Plugins
The following produces a 440Hz sine tone (e.g. for tuning purposes) and plays it out with the default ALSA PCM device. For this to work, one needs to have the LADSPA SDK needs installed (see www.ladspa.org).
1. ecasound -i null -o alsa -el:sine_fcac,440,1
The next example produces a metrome signal with tempo of 120BPM, using a LADSPA plugin to create the sine tone (-el), ecasound pulse gate (-eemb) to create the pulse and finally a low-pass filter (-efl) to filter the resulting output.
2. ecasound -i:null -o:alsa -el:sine_fcac,880,1 -eemb:120,10 -efl:2000
LV2 Effect Plugins
LV2 is a plugin standard for audio systems, designed as a successor for LADSPA. See lv2plug.in website for more information, and links to available LV2 plugins, documentation and other resources.In the first example, a +6dB gain is applied using LV2 example 'eq-amp' plugin. LV2 plugins are identified with URIs, so the first parameter to '-elv2' is the plugin URI (e.g. http://lv2plug.in/plugins/eg-amp).
1. ecasound -i foobar.wav -o alsa -elv2:http://lv2plug.in/plugins/eg-amp,6
The next example shows how to use the analogueOsc plugin from swh-plugins package, to create a 400Hz square tone and play it to default ALSA audio output:
LV2 (like LADSPA) plugins can be controlled just like other Ecasound plugins. E.g. making a 10sec 20-2000Hz sweep of the square tone from previous example:2. ecasound -i:null -o:alsa -elv2:http://plugin.org.uk/swh-plugins/analogueOsc,3,400,0.4,0.05
3. ecasound -i:null -o:alsa -elv2:http://plugin.org.uk/swh-plugins/analogueOsc,3,20,0.4,0.05 -kl:2,20,2000,10
Version note: For LV2 support, you need ecasound 2.9.0 or newer.
In the above example for LADSPA, the sequence
-el:sine_fcac,880,1 -eemb:120,10 -efl:2000
was used to create a metronome signal. Combinations of effects like this can
be stored for later use as Ecasound presets:
This is defined in @prefix@/share/ecasound/effect_presets as:1. ecasound -i:null -o:alsa -pn:metronome,120
# the metronome example, taking a single parameter (tempo) metronome = -el:sine_fcac,880,1 -eemb:%1,10 -efl:2000 \ -ppn:bpm -ppd:100 -pd:Sineosc_metronome
Passing parameters is supported (%1 is the first preset parameter, %2 the second, and so forth). In the above example, %1 is set to 120 and this may be modified during runtime (interactively, with oscillators, via MIDI, OSC, or any other method available for controlling parameters).
A sample 'effect_presets' file is distributed with Ecasound.
More MIDI examples
The following is a simple example how an MIDI continuous controller (CC) is used to control an ecasound effect parameter (the cutoff frequency of the low-pass filter '-efl'):
1. ecasound -i somefile.wav -o alsa -efl:400 -km:1,400,4200,74,1
The above example uses OSS rawmidi access, which is equivalent to adding '-Md:rawmidi,/dev/midi' to the command line. To do the same with ALSA, the syntax is as follows:
2. ecasound -i somefile.wav -o alsa -efl:400 -km:1,400,4200,74,1 -Md:rawmidi,/dev/snd/midiC0D0 3. ecasound -i somefile.wav -o alsa -efl:400 -km:1,400,4200,74,1 -Md:alsaseq,80:1 4. ecasound -i somefile.wav -o alsa -efl:400 -km:1,400,4200,74,1 -Md:alsaseq,KMidimon
The first one uses the ALSA rawmidi interface and opens an ALSA device (card 0, device 0) for raw MIDI I/0. The second example uses the ALSA sequencer API, which is more powerful as you can route MIDI packets not only to hardware interfaces, but also to and from other applications supporting the sequencer API. "80,1" and "KMidimon" are sequencr ports to which ecasound should connect. You can use the "aconnect" (part of the alsa-utils package) tool to list all available sequencer ports.
Version note: For ALSA sequencer support, you need ecasound 2.4.3 or newer.
Ecasound also provides a few tone generation primitives. These are mainly intended for testing, but can have other uses as well. For a more versatile set of tone generators, one can utilize the various LADSPA plugins that produce tones.
A sine tone of 880Hz, played out to the default ALSA sound device:
1. ecasound -i tone,sine,880 -o alsa
The above continuous to produce the tone indefinitely. It is also possible to create a test tone of finite length. First, a sine tone of 440Hz with length of exactly 22.25secs. Then a similar example, in which the length of the tone is given in samples (88200 in this case):
2. ecasound -i tone,sine,440,22.25 -o alsa 3. ecasound -i tone,sine,880,88200sa -o alsa
Version note: 'tone' audio object type was added to ecasound version 2.5.0.
Piping data from other processes
Here is one example of how audio can be piped to ecasound:
1. mpg123 -s sometune.mp3 | ecasound -i:stdin -o alsa
The above sends the output of mpg123 to standard output ('-s' option of mpg123) and reads it from standard input with ecasound ('-i:stdin' option), and plays it out through the default ALSA PCM device. Similarly, 'stdout' can be used to pipe data out from ecasound.
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