lpread — Reads a control file of time-ordered information frames.
ifilcod -- integer or character-string denoting a control-file (reflection coefficients and four parameter values) derived from n-pole linear predictive spectral analysis of a source audio signal. An integer denotes the suffix of a file lp.m; a character-string (in double quotes) gives a filename, optionally a full pathname. If not fullpath, the file is sought first in the current directory, then in that of the environment variable SADIR (if defined). Memory usage depends on the size of the file, which is held entirely in memory during computation but shared by multiple calls (see also adsyn, pvoc).
inpoles (optional, default=0) -- number of poles in the lpc analysis. It is required only when the control file does not have a header; it is ignored when a header is detected.
ifrmrate (optional, default=0) -- frame rate per second in the lpc analysis. It is required only when the control file does not have a header; it is ignored when a header is detected.
lpread accesses a control file of time-ordered information frames, each containing n-pole filter coefficients derived from linear predictive analysis of a source signal at fixed time intervals (e.g. 1/100 of a second), plus four parameter values:
krmsr -- root-mean-square (rms) of the residual of analysis
krmso -- rms of the original signal
kerr -- the normalized error signal
kcps -- pitch in Hz
ktimpnt -- The passage of time, in seconds, through the analysis file. ktimpnt must always be positive, but can move forwards or backwards in time, be stationary or discontinuous, as a pointer into the analysis file.
lpread gets its values from the control file according to the input value ktimpnt (in seconds). If ktimpnt proceeds at the analysis rate, time-normal synthesis will result; proceeding at a faster, slower, or variable rate will result in time-warped synthesis. At each k-period, lpread interpolates between adjacent frames to more accurately determine the parameter values (presented as output) and the filter coefficient settings (passed internally to a subsequent lpreson).
The error signal kerr (between 0 and 1) derived during predictive analysis reflects the deterministic/random nature of the analyzed source. This will emerge low for pitched (periodic) material and higher for noisy material. The transition from voiced to unvoiced speech, for example, produces an error signal value of about .3. During synthesis, the error signal value can be used to determine the nature of the lpreson driving function: for example, by arbitrating between pitched and non-pitched input, or even by determining a mix of the two. In normal speech resynthesis, the pitched input to lpreson is a wideband periodic signal or pulse train derived from a unit such as buzz, and the nonpitched source is usually derived from rand. However, any audio signal can be used as the driving function, the only assumption of the analysis being that it has a flat response.
lpfreson is a formant shifted lpreson, in which kfrqratio is the (cps) ratio of shifted to original formant positions. This permits synthesis in which the source object changes its apparent acoustic size. lpfreson with kfrqratio = 1 is equivalent to lpreson.
Generally, lpreson provides a means whereby the time-varying content and spectral shaping of a composite audio signal can be controlled by the dynamic spectral content of another. There can be any number of lpread/lpreson (or lpfreson) pairs in an instrument or in an orchestra; they can read from the same or different control files independently.
Here is an example of the lpread opcode. It uses the file lpread.csd.
Example 459. Example of the lpread opcode.
See the sections Real-time Audio and Command Line Flags for more information on using command line flags.
<CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o lpread.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 instr 1 ; works with or without -a option when analyzing "fox.wav" from the manual ilen filelen "fox.wav" ; length of soundfile prints "fox.wav = %f seconds\\n",ilen ktime line 0, p3, p4 krmsr,krmso,kerr,kcps lpread ktime,"fox_poles.lpc" krmso = krmso*.00007 ; low volume aout buzz krmso, kcps, 15, 1 krmsr = krmsr*.0001 ; low volume asig rand krmsr outs (aout*2)+asig, (aout*2)+asig ; mix buzz and rand endin </CsInstruments> <CsScore> ; sine f1 0 4096 10 1 i 1 0 2.8 1 ; first words only i 1 4 2.8 2.8 ; whole sentence e </CsScore> </CsoundSynthesizer>
The audio file “fox.wav” is 2.8 seconds long. So filelen's output should be a line like this:
fox.wav = 2.756667 seconds