cvanal -- converts a soundfile into a single Fourier transform frame. The output file can be used by the convolve operator to perform Fast Convolution between an input signal and the original impulse response. Analysis is conditioned by the flags below. A space is optional between the flag and its argument.
-s rate -- 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. If omitted, the default is to process all channels. If a value is given, only the selected channel will be processed.
-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.
-X -- write machine independent format analysis file.
The output file has a special convolve header, containing details of the source audio file. The analysis data is stored as “float”, in rectangular (real/imaginary) form.
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The analysis file is not system independent! Ensure that the original impulse recording/data is retained. If/when required, the analysis file can be recreated. |
cvanal asound cvfile
will analyze the soundfile "asound" to produce the file "cvfile" for the use with convolve
To use data that is not already contained in a soundfile, a soundfile converter that accepts text files may be used to create a standard audio file, e.g., the .DAT format for SOX. This is useful for implementing FIR filters.
Here is an example of the cvanal utility. Play cvanal.csd
Example 1353. Example of the cvanal utility.
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 -m0 --limiter=.99 ;;;realtime audio out, with limiter protection ; For Non-realtime ouput leave only the line below: ; -o cvanal.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ; by Menno Knevel 2021 gilen filelen "rv_stereo.wav" ; get length of impulse soundfile ; analyze sound file and output result to 3 convolve files ires1 system_i 1,{{ cvanal rv_stereo.wav rv_stereo1.con }} ; default settings ires2 system_i 1,{{ cvanal -d.15 rv_stereo.wav rv_stereo2.con }} ; use only first portion ires3 system_i 1,{{ cvanal -b.25 -d.001 rv_stereo.wav rv_stereo3.con }} ; take very short portion from the end instr 1 ; untreated signal asig diskin2 "beats.wav", 1 prints "\n---***YOU NOW HEAR THE UNTREATED SOUND SAMPLE***---\n" outs asig, asig endin instr 2 prints "\n---***YOU NOW HEAR THE RESULT OF THIS ANALYZED FILE:***---\n" prints "--**used duration of impulse file = %5.3f seconds (total =%5.2f)**--\\n", p5, gilen adry diskin2 "beats.wav" ; input (dry) audio awet1, awet2 convolve adry*.8, p4 ; stereo convolved (wet) audio awet1 *= p6 ; scale amplitude of impulse sound, Left awet2 *= p6 ; & Right channel adrydel delay adry, p5 ; delay dry signal to align it with convolved signal outs (adrydel+awet1)*.8,(adrydel+awet2)*.8 ; mix wet & dry signals endin </CsInstruments> <CsScore> i1 1 2 ; untreated signal i2 4 3 "rv_stereo1.con" 0.39 .01 ; use total impulse i2 8 3 "rv_stereo2.con" 0.15 .01 ; first portion of impulse i2 12 3 "rv_stereo3.con" 0.001 1 ; very short portion of impulse, starting from 0.25 sec, & scale amp e </CsScore> </CsoundSynthesizer>