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|
#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
|
