12-13-2010
Please show sample input and desired output for us to get the exact idea about your problem.
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Input file :
ID1~Name1~Place1
ID2~Name2~Place2
ID3~Name3~Place3
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ID1<<12 spaces>>Name1~Place1
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8 8
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Pval.txt
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91500 0.004
91700 0.007
91800 0.12
91900 0.05
92000 0.99
Freq.txt
ID Frequency
90000 56
91500 29
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Match column 3 in file1 to column 1 in file 2 and replace with column 2 from file2
file 1 sample
SNDK 80004C101 AT
XLNX 983919101 BB
NETL 64118B100 BS
AMD 007903107 CC
KLAC 482480100 DC
TER 880770102 KATS
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Hi all,
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File1
ABA
ABC
ABE
ABF
File 2
ABA 123
ABB 124
ABD 125
ABC 126
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Hi,
I have two input files as
File1 :
ABC:client1:project1
XYZ:client2-aa:project2
DEF:client4:proj
File2 :
client1:W-170:xx
client2-aa:WT-04:yy
client4:L-005A:zz
Also, array of valid values can be hardcoded like
Output :
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Hi,
I have fileA.txt like this.
B01B02 D0011718
B01B03 D0012540
B01B04 D0006145
B01B05 D0004815
B01B06 D0012069
B01B07 D0004064
B01B08 D0011988
B01B09 D0012071
B01B10 D0005596
B01B11 D0011351
B01B12 D0004814
B01C01 D0011804
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LEARN ABOUT DEBIAN
srconv
SRCONV(1) The Canonical Csound Reference SRCONV(1)
NAME
srconv - Converts the sample rate of an audio file. .
DESCRIPTION
Converts the sample rate of an audio file at sample rate Rin to a sample rate of Rout. Optionally the ratio (Rin / Rout) may be linearly
time-varying according to a set of (time, ratio) pairs in an auxiliary file.
SYNTAX
srconv [flags] infile
INITIALIZATION
Flags:
o -P num = pitch transposition ratio (srate / r) [don't specify both P and r]
o -P num = pitch transposition ratio (srate / r) [don't specify both P and r]
o -Q num =quality factor (1, 2, 3, or 4: default = 2)
o -i filnam = auxiliary breakpoints file (no breakpoint by default. i.e. No ratio change)
o -r num = output sample rate (must be specified)
o -o fnam = sound output filename
o -A = create an AIFF format output soundfile
o -J = create an IRCAM format output soundfile
o -W = create a WAV format output soundfile
o -h = no header on output soundfile
o -c = 8-bit signed_char sound samples
o -a = alaw sound samples
o -8 = 8-bit unsigned_char sound samples
o -u = ulaw sound samples
o -s = short_int sound samples
o -l = long_int sound samples
o -f = float sound samples
o -r N = orchestra srate override
o -K = Do not generate PEAK chunks
o -R = continually rewrite header while writing soundfile (WAV/AIFF)
o -H# = print a heartbeat style 1, 2 or 3 at each soundfile write
o -N = notify (ring the bell) when score or miditrack is done
o -- fnam = log output to file
This program performs arbitrary sample-rate conversion with high fidelity. The method is to step through the input at the desired sampling
increment, and to compute the output points as appropriately weighted averages of the surrounding input points. There are two cases to
consider:
1. sample rates are in a small-integer ratio - weights are obtained from table.
2. sample rates are in a large-integer ratio - weights are linearly interpolated from table.
Calculate increment: if decimating, then window is impulse response of low-pass filter with cutoff frequency at half of output sample rate;
if interpolating, then window is impulse response of lowpass filter with cutoff frequency at half of input sample rate.
CREDITS
Author: Mark Dolson
August 26, 1989
Author: John ffitch
December 30, 2000
AUTHORS
Barry Vercoe
MIT Media Lab
Author.
Dan Ellis
MIT Media Lab,
Cambridge
Massachussetts
Author.
COPYRIGHT
5.10 08/01/2011 SRCONV(1)