Edit File: _ecasound
#compdef ecasound local curcontext="$curcontext" state line expl typeset -A opt_args _arguments \ '-c[start in interactive mode]' \ '(-q)-d\:-[debug level]:debug level' \ '-D[print all debug information to stderr]' \ '(-d)-q[quiet mode, no output]' \ '(-)--help[show usage information]' \ '(-)--version[show version information]' \ '-n\:-[set the name of chainsetup]:chainsetup name' \ '-s\:-[create a new chainsetup from file]:chainsetup file:_files' \ '-sr\:-[set internal sampling rate]:internal sampling rate:(8000 11025 22050 44100 48000)' \ '*-a\:-[select active signal chains]:chain name' \ '-b\:-[set the size of buffer in samples]:buffer size:->b' \ '-m\:-[force use of specified mix mode]:mix mode:((auto\:Automatic simple\:Only\ one\ input/cain/output normal\:Normal\ single-threaded\ mode))' \ '-r[use realtime scheduling policy (SCHED_FIFO)]' \ '-r\:-[use realtime scheduling policy (SCHED_FIFO)]:sched_priority' \ '-x[truncate outputs]' \ '*-z\:-[enable feature]:feature:->z' \ '-t\:-[set processing time in seconds]:seconds (int/float)' \ '-tl[enable looping]' \ '*-f\:-[set sampling parameters for the following input/output files]: :->f' \ '*-y\:-[set starting position for last specified input/output]:seconds' \ '*-i\:-[specifies a new input source]:input source:->io' \ '*-o\:-[specifies a new output source]:output source:->io' \ '*-Md\:-[set the active MIDI-device]:device name:_files' \ '*-Mms\:-[send MMC start/stop to MIDI device-id]:device id' \ '*-mss[sends MIDI-sync to the selected MIDI-device]' \ '*-pf\:-[use the first preset found from file as chain operator]:preset file:_files -g \*.epp\(-.\)' \ '*-pn\:-[find preset from global preset database]:preset name:->pn' \ '*-ev[analyze sample data to find max apm value without clipping]' \ '*-ezf[find the optimal value for DC-adjusting]' \ '*-eS\:-[audio stamp]:stamp-id (int)' \ '*-ea\:-[amplify signal]:amplification value (percent)' \ '*-eac\:-[amplify signal of channel]: :->eac' \ '*-eaw\:-[amplify singal (clipping)]: :->eaw' \ '*-eal\:-[limits audio level]:limit (percent)' \ '*-ec\:-[compressor (a simple one)]: :->ec' \ '*-eca\:-[a more advanced compressor]: :->eca' \ '*-enm\:-[noise gate. (each channel is processes separately)]: :->enm' \ '*-ei\:-[pitch shifter (modifies audio pitch by altering its length)]:pitch-shift (percent)' \ '*-epp\:-[normal pan effect]:panning (0=left, 50=center, 100=right)' \ '*-ezx\:-[adjusts the signal DC (use -ezf to find optimal values)]: :->ezx' \ '*-eem-[envelope modulation]: :->emod' \ '*-ef-[apply filter effects]: :->filters' \ '*-erc\:-[copy channel]: :->erc' \ '*-erm\:-[mix all channels to one channel]:to channel' \ '*-et-[time based effects]: :->teffects' \ '*-el\:-[LADSPA Plugin]: :->el' \ '*-eli\:-[LADSPA Plugin]: :->el' \ '*-gc\:-[time crop gate]: :->gc' \ '*-ge\:-[threshold gate]: :->ge' \ && return 0 case $state in filters) _values -S : 'filter effect' \ '1[resonant bandpass filter]: :->ef1' \ '3[resonant lowpass filter]: :->ef3' \ '4[resonant lowpass filter (3rd-order, 36dB)]: :->ef4' \ 'a[allpass filter]: :->efa' \ 'c[comb filter]: :->efc' \ 'b[bandpass filter]: :->efb' \ 'h[highpass filter]:cutoff frequency' \ 'i[inverse comb filter]: :->efi' \ 'l[lowpass filter]:cutoff frequency' \ 'r[bandreject filter]: :->efr' \ 's[resonator (resonating bandpass filter)]: :->efs' ;; teffects) _values -S : 'time based effect' \ 'c[chorus]: :->etc' \ 'd[delay effect]: :->etd' \ 'e[a more advanced reverb effect]: :->ete' \ 'f[fake-stereo effect]:delay time (msec)' \ 'l[flanger]: :->etl' \ 'm[multitap delay]: :->etm' \ 'p[phaser]: :->etp' \ 'r[reverb effect]: :->etr' ;; emod) _values -S : 'envelopme modulation' \ 'b[pulse gate]: :->eemb' \ 'p[pulse gate (hz)]: :->eemp' \ 't[tremolo effect]: :->eemt' ;; esac case $state in b) _wanted -V sizes expl 'buffer size' compadd \ 1 2 4 8 16 32 64 128 256 512 1024 2048 4096 8192 16384 32768 65536 ;; f) if compset -P '*,*,*,'; then _values 'interleaving' \ 'i[interleaved stream format]' \ 'n[noninterleaved]' elif compset -P '*,*,'; then _message 'sampling rate' elif compset -P '*,'; then _message 'channels' else _values 'sampling parameters' \ 'u8[unsigned 8-bit]' \ 's16_le[signed 16-bit little endian]' \ 's16_be[signed 16-bit big endian]' \ 's24_le[signed 24-bit little endian]' \ 's24_be[signed 24-bit big endian]' \ 's32_le[signed 32-bit little endian]' \ 's32_be[signed 32-bit big endian]' \ 'f32_le[32-bit float (little endian)]' \ 'f32_be[32-bit float (big endian)]' fi ;; z) _values -s , feature \ '(nodb)db[enable double-buffering]' \ '(db)nodb[disable double-buffering]' \ 'dbsize[set db buffer size]:buffer size in sample frames:(0 1 2 4 8 16)' \ '(nointbuf)intbuf[use extra internal buffering for realtime devices]' \ '(intbuf)nointbuf[prevent extra internal buffering for realtime devices]' \ 'xruns[processing will be halted when a under/overrun occurs]' \ 'psr[enable the precise-sample-rates]' ;; io) if compset -P 'alsa,'; then if [[ -e /proc/asound ]]; then eval `grep "^[[:digit:]]" < /proc/asound/cards|awk 'BEGIN {print "_values '\''ALSA device'\''" }; {print "'\''" $1 "[" $6, $7, $8, $9 "]'\''"}'||echo _message Wrong` else _message 'ALSA information bot found in proc filesystem' fi else _alternative \ 'files:input/output file:_files -g "*.(aif|aiff|mid|wav|ewf|mp3|mp2)(-.)"' \ 'streams:stream:(stdin stdout)' \ 'devices:realtime device:((/dev/dsp alsa\:alsa\ device null\:null\ device))' fi ;; pn) _wanted presets expl preset compadd \ ${${(M)${(f)"$(</usr/share/ecasound/effect_presets)"}:#[a-z]*}%% *} ;; etd) if compset -P 3 '*,'; then _message 'mix (wet) (percent)' elif compset -P 2 '*,'; then _message 'number of delays' elif compset -P '*,'; then _values -s , 'surround mode' \ '0[normal]' \ '1[surround]' \ '2[stereo-spread]' else _message 'delay time (msec)' fi ;; ge) if compset -P 2 '*,'; then _values 'volume mode' 'rms' elif compset -P '*,'; then _message 'close threshold (percent)' else _message 'open threshold (percent)' fi ;; *) # all the rest are comma separated lists for which we use a message prompt typeset -A msgs local str msgs=( eac 'amplification value (percent):channel' eaw 'amplification value (percent):channel:max-clipped-samples' ec 'compression rate (decibels):threshold (0.0-1.0)' eca 'peak-level:release-time (seconds):fast compression rate (0.0-1.0):compression rate' enm 'threshold-level:pre-hold-time (msec):attack-time (msec):post-hold-time (msec):release-time (msec)' ezx 'left DC fix value:right CD fix value' eemb 'pulse frequency (beats per minute):on time (msec)' eemp 'pulse frequency (hz):on time (percent)' eemt 'tremolo speed (beats per second):depth (percent)' ef1 'center frequency:width (Hz)' ef3 'cutoff frequency:resonance:gain' ef4 'cutoff:resonance' efa 'delay (samples):feedback (percent)' efc 'delay (samples):radius (0-1.0)' efb 'center frequency:width (Hz)' efi 'delay (samples):radius (0-1.0)' efr 'center frequency:width (Hz)' efs 'center frequency:width (Hz)' erc 'from channel:to channel' etc 'delay time (msec):variance time (samples):feedback (percent):lfo frequency' ete 'room size (metres):feedback level (percent):amount of reverbed signal added to the original (wet) (percent)' etl 'delay time (msec):variance time (samples):feedback (percent):lfo frequency' etm 'delay time (msec):number of delays:mix (wet) (percent)' etp 'delay time (msec):variance time (samples):feedback (percent):lfo frequency' etr 'delay time (msec):surround mode:feedback (percent)' el 'unique LADSPA name/number' gc 'start time (seconds):length (seconds)' ) str=$msgs[$state] while compset -P 1 '*,'; do str="${str#*:}" done _message "${str%%:*}" ;; esac
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