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