Linux and UNIX Man Pages

Linux & Unix Commands - Search Man Pages

lpanal(1) [linux man page]

LPANAL(1)						  The Canonical Csound Reference						 LPANAL(1)

NAME
lpanal - Performs both linear predictive and pitch-tracking analysis on a soundfile. . DESCRIPTION
Linear predictive analysis for the Csound Linear Predictive Coding (LPC) Resynthesis opcodes. SYNTAX
csound -U lpanal [flags] infilename outfilename lpanal [flags] infilename outfilename INITIALIZATION
lpanal performs both lpc and pitch-tracking analysis on a soundfile to produce a time-ordered sequence of frames of control information suitable for Csound resynthesis. Analysis is conditioned by the control flags below. A space is optional between the flag and its value. -a -- [alternate storage] asks lpanal to write a file with filter poles values rather than the usual filter coefficient files. When lpread / lpreson are used with pole files, automatic stabilization is performed and the filter should not get wild. (This is the default in the Windows GUI) - Changed by Marc Resibois. -s srate -- sampling rate of the audio input file. This will over-ride the srate of the soundfile header, which otherwise applies. If neither is present, the default is 10000. -c channel -- channel number sought. The default is 1. -b begin -- beginning time (in seconds) of the audio segment to be analyzed. The default is 0.0 -d duration -- duration (in seconds) of the audio segment to be analyzed. The default of 0.0 means to the end of the file. -p npoles -- number of poles for analysis. The default is 34, the maximum 50. -h hopsize -- hop size (in samples) between frames of analysis. This determines the number of frames per second (srate / hopsize) in the output control file. The analysis framesize is hopsize * 2 samples. The default is 200, the maximum 500. -C string -- text for the comments field of the lpfile header. The default is the null string. -P mincps -- lowest frequency (in Hz) of pitch tracking. -P0 means no pitch tracking. -Q maxcps -- highest frequency (in Hz) of pitch tracking. The narrower the pitch range, the more accurate the pitch estimate. The defaults are -P70, -Q200. -v verbosity -- level of terminal information during analysis. o 0 = none o 1 = verbose o 2 = debug The default is 0. EXAMPLES
lpanal -a -p26 -d2.5 -P100 -Q400 audiofile.test lpfil22 will analyze the first 2.5 seconds of file "audiofile.test", producing srate/200 frames per second, each containing 26-pole filter coefficients and a pitch estimate between 100 and 400 Hertz. Stabilized (-a) output will be placed in "lpfil22" in the current directory. File Format Output is a file comprised of an identifiable header plus a set of frames of floating point analysis data. Each frame contains four values of pitch and gain information, followed by npoles filter coefficients. The file is readable by Csound's lpread. lpanal is an extensive modification of Paul Lanksy's lpc analysis programs. AUTHORS
Barry Vercoe MIT Media Lab Author. Dan Ellis MIT Media Lab, Cambridge Massachussetts Author. COPYRIGHT
5.07 06/23/2009 LPANAL(1)

Check Out this Related Man Page

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
5.07 06/23/2009 SRCONV(1)
Man Page