gr_plot_fft_c(1) [debian man page]
GR_PLOT_FFT_C(1) User Commands GR_PLOT_FFT_C(1) NAME
gr_plot_fft_c - plot complex binary data using GNU Radio SYNOPSIS
gr_plot_fft_c: [options] input_filename DESCRIPTION
Takes a GNU Radio complex binary file and displays the I&Q data versus time as well as the frequency domain (FFT) plot. The y-axis values are plotted assuming volts as the amplitude of the I&Q streams and converted into dBm in the frequency domain (the 1/N power adjustment out of the FFT is performed internally). The script plots a certain block of data at a time, specified on the command line as -B or --block. This value defaults to 1000. The start position in the file can be set by specifying -s or --start and defaults to 0 (the start of the file). By default, the system assumes a sample rate of 1, so in time, each sample is plotted versus the sample number. To set a true time and frequency axis, set the sample rate (-R or --sample-rate) to the sample rate used when capturing the samples. OPTIONS
-h, --help show this help message and exit -B BLOCK, --block=BLOCK Specify the block size [default=1000] -s START, --start=START Specify where to start in the file [default=0] -R SAMPLE_RATE, --sample-rate=SAMPLE_RATE Set the sampler rate of the data [default=1.0] SEE ALSO
gr_plot_char(1) gr_plot_const(1) gr_plot_fft_c(1) gr_plot_fft_f(1) gr_plot_float(1) gr_plot_int(1) gr_plot_iq(1) gr_plot_psd_c(1) gr_plot_psd_f(1) gr_plot_qt(1) gr_plot_short(1) gr_plot_fft_c 3.5 December 2011 GR_PLOT_FFT_C(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
5.10 08/01/2011 SRCONV(1)