kopia lustrzana https://github.com/ha7ilm/csdr
Retabbed most source files to 4 spaces
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README.md
182
README.md
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@ -13,8 +13,8 @@ Most of the code is available under the permissive BSD license, with some option
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How to compile
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--------------
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make
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sudo make install
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make
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sudo make install
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The project was only tested on Linux. It has the following dependencies: `libfftw3-dev`
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@ -24,7 +24,7 @@ To run the examples, you will also need <a href="http://sdr.osmocom.org/trac/wik
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If you compile *fftw3* from sources for use with *libcsdr*, you need to configure it with 32-bit float support enabled:
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./configure --enable-float
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./configure --enable-float
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(This is for *fftw3*, not *libcsdr*. You do not need to run the configure script before compiling *libcsdr*.)
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@ -39,7 +39,7 @@ Usage by example
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### Demodulate WFM
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rtl_sdr -s 240000 -f 89500000 -g 20 - | csdr convert_u8_f | csdr fmdemod_quadri_cf | csdr fractional_decimator_ff 5 | csdr deemphasis_wfm_ff 48000 50e-6 | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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rtl_sdr -s 240000 -f 89500000 -g 20 - | csdr convert_u8_f | csdr fmdemod_quadri_cf | csdr fractional_decimator_ff 5 | csdr deemphasis_wfm_ff 48000 50e-6 | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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- Baseband I/Q signal is coming from an RTL-SDR USB dongle, with a center frequency of `-f 104300000` Hz, a sampling rate of `-s 240000` samples per second.
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- The `rtl_sdr` tool outputs an unsigned 8-bit I/Q signal (one byte of I sample and one byte of Q coming after each other), but `libcsdr` DSP routines internally use floating point data type, so we convert the data stream of `unsigned char` to `float` by `csdr convert_u8_f`.
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@ -51,7 +51,7 @@ Usage by example
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### Demodulate WFM: advanced
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rtl_sdr -s 2400000 -f 89300000 -g 20 - | csdr convert_u8_f | csdr shift_addition_cc -0.085 | csdr fir_decimate_cc 10 0.05 HAMMING | csdr fmdemod_quadri_cf | csdr fractional_decimator_ff 5 | csdr deemphasis_wfm_ff 48000 50e-6 | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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rtl_sdr -s 2400000 -f 89300000 -g 20 - | csdr convert_u8_f | csdr shift_addition_cc -0.085 | csdr fir_decimate_cc 10 0.05 HAMMING | csdr fmdemod_quadri_cf | csdr fractional_decimator_ff 5 | csdr deemphasis_wfm_ff 48000 50e-6 | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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- We want to listen to one radio station, but input signal contains multiple stations, and its bandwidth is too large for sending it directly to the FM demodulator.
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- We shift the signal to the center frequency of the station we want to receive: `-0.085*2400000 = -204000`, so basically we will listen to the radio station centered at 89504000 Hz.
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@ -60,19 +60,19 @@ Usage by example
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Sample rates look like this:
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2.4 Msps 240 ksps 48 ksps
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I/Q source ------------> FIR decimation ------------> FM demod -> frac. decimation ---------> deemphasis -> sound card
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2.4 Msps 240 ksps 48 ksps
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I/Q source ------------> FIR decimation ------------> FM demod -> frac. decimation ---------> deemphasis -> sound card
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*Note:* there is an example shell script that does this for you (without the unnecessary shift operation). If you just want to listen to FM radio, type:
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csdr-fm 89.5 20
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csdr-fm 89.5 20
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The first parameter is the frequency in MHz, and the second optional parameter is the RTL-SDR tuner gain in dB.
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### Demodulate NFM
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rtl_sdr -s 2400000 -f 145000000 -g 20 - | csdr convert_u8_f | csdr shift_addition_cc `python -c "print float(145000000-145350000)/2400000"` | csdr fir_decimate_cc 50 0.005 HAMMING | csdr fmdemod_quadri_cf | csdr limit_ff | csdr deemphasis_nfm_ff 48000 | csdr fastagc_ff | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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rtl_sdr -s 2400000 -f 145000000 -g 20 - | csdr convert_u8_f | csdr shift_addition_cc `python -c "print float(145000000-145350000)/2400000"` | csdr fir_decimate_cc 50 0.005 HAMMING | csdr fmdemod_quadri_cf | csdr limit_ff | csdr deemphasis_nfm_ff 48000 | csdr fastagc_ff | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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- Note that the decimation factor is higher (we want to select a ~25 kHz channel).
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- Also there is a python hack to calculate the relative shift offset. The real receiver frequency is `145350000` Hz.
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@ -80,14 +80,14 @@ The first parameter is the frequency in MHz, and the second optional parameter i
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### Demodulate AM
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rtl_sdr -s 2400000 -f 145000000 -g 20 - | csdr convert_u8_f | csdr shift_addition_cc `python -c "print float(145000000-144400000)/2400000"` | csdr fir_decimate_cc 50 0.005 HAMMING | csdr amdemod_cf | csdr fastdcblock_ff | csdr agc_ff | csdr limit_ff | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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rtl_sdr -s 2400000 -f 145000000 -g 20 - | csdr convert_u8_f | csdr shift_addition_cc `python -c "print float(145000000-144400000)/2400000"` | csdr fir_decimate_cc 50 0.005 HAMMING | csdr amdemod_cf | csdr fastdcblock_ff | csdr agc_ff | csdr limit_ff | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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- `amdemod_cf` is used as demodulator.
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- `agc_ff` should be used for AM and SSB.
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### Design FIR band-pass filter (with complex taps)
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csdr firdes_bandpass_c 0 0.5 59 HAMMING --octave | octave -i
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csdr firdes_bandpass_c 0 0.5 59 HAMMING --octave | octave -i
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- ...and then plot its frequency response with octave. (You can close octave window by issuing Ctrl-C in the terminal window.)
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- It will design a filter that lets only the positive frequencies pass (low cut is 0, high cut is 0.5 - these are relative to the sampling rate).
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@ -95,13 +95,13 @@ The first parameter is the frequency in MHz, and the second optional parameter i
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### Demodulate SSB
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rtl_sdr -s 2400000 -f 145000000 -g 20 - | csdr convert_u8_f | csdr shift_addition_cc `python -c "print float(145000000-144400000)/2400000"` | csdr fir_decimate_cc 50 0.005 HAMMING | csdr bandpass_fir_fft_cc 0 0.1 0.05 | csdr realpart_cf | csdr agc_ff | csdr limit_ff | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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rtl_sdr -s 2400000 -f 145000000 -g 20 - | csdr convert_u8_f | csdr shift_addition_cc `python -c "print float(145000000-144400000)/2400000"` | csdr fir_decimate_cc 50 0.005 HAMMING | csdr bandpass_fir_fft_cc 0 0.1 0.05 | csdr realpart_cf | csdr agc_ff | csdr limit_ff | csdr convert_f_s16 | mplayer -cache 1024 -quiet -rawaudio samplesize=2:channels=1:rate=48000 -demuxer rawaudio -
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- It is a modified Weaver-demodulator. The complex FIR filter removes the lower sideband and lets only the upper pass (USB). If you want to demodulate LSB, change `bandpass_fir_fft_cc 0 0.05` to `bandpass_fir_fft_cc -0.05 0`.
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### Draw FFT
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rtl_sdr -s 2400000 -f 104300000 -g 20 - | csdr convert_u8_f | csdr fft_cc 1024 1200000 HAMMING --octave | octave -i > /dev/null
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rtl_sdr -s 2400000 -f 104300000 -g 20 - | csdr convert_u8_f | csdr fft_cc 1024 1200000 HAMMING --octave | octave -i > /dev/null
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- We calculate the Fast Fourier Transform by `csdr fft_cc` on the first 1024 samples of every block of 1200000 complex samples coming after each other. (We calculate FFT from 1024 samples and then skip 1200000-1024=1198976 samples. This way we will calculate FFT two times every second.)
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- The window used for FFT is the Hamming window, and the output consists of commands that can be directly interpreted by GNU Octave which plots us the spectrum.
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@ -153,7 +153,7 @@ Optional parameters have safe defaults, for more info look at the code.
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Syntax:
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csdr realpart_cf
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csdr realpart_cf
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It takes the real part of the complex signal, and throws away the imaginary part.
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@ -163,7 +163,7 @@ It takes the real part of the complex signal, and throws away the imaginary part
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Syntax:
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csdr clipdetect_ff
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csdr clipdetect_ff
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It clones the signal (the input and the output is the same), but it prints a warning on `stderr` if any sample value is out of the -1.0 ... 1.0 range.
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@ -173,7 +173,7 @@ It clones the signal (the input and the output is the same), but it prints a war
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Syntax:
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csdr limit_ff [max_amplitude]
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csdr limit_ff [max_amplitude]
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The input signal amplitude will not be let out of the `-max_amplitude ... max_amplitude` range.
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@ -183,7 +183,7 @@ The input signal amplitude will not be let out of the `-max_amplitude ... max_am
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Syntax:
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csdr gain_ff <gain>
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csdr gain_ff <gain>
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It multiplies all samples by `gain`.
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@ -193,7 +193,7 @@ It multiplies all samples by `gain`.
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Syntax:
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csdr clone
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csdr clone
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It copies the input to the output.
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@ -203,7 +203,7 @@ It copies the input to the output.
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Syntax:
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csdr through
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csdr through
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It copies the input to the output, while also displaying the data rate going through it.
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@ -213,7 +213,7 @@ It copies the input to the output, while also displaying the data rate going thr
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Syntax:
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csdr none
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csdr none
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The `csdr` process just exits with 0.
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@ -223,7 +223,7 @@ The `csdr` process just exits with 0.
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Syntax:
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csdr yes_f <to_repeat> [buf_times]
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csdr yes_f <to_repeat> [buf_times]
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It outputs continously the `to_repeat` float number.
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@ -237,7 +237,7 @@ Else, after outputing `buf_times` number of buffers (the size of which is stated
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Syntax:
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csdr detect_nan_ff
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csdr detect_nan_ff
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Along with copying its input samples to the output, it prints a warning message to *stderr* if it finds any IEEE floating point NaN values among the samples.
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@ -247,7 +247,7 @@ Along with copying its input samples to the output, it prints a warning message
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Syntax:
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csdr dump_f
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csdr dump_f
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It prints all floating point input samples as text.
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@ -265,7 +265,7 @@ Aliases for this function: `floatdump_f`
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Syntax:
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csdr dump_u8
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csdr dump_u8
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It prints all input bytes as text, in hexadecimal form.
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@ -277,7 +277,7 @@ Alternatively, you can use the `xxd` command (built into most Linux distribution
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Syntax:
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csdr flowcontrol <data_rate> <reads_per_second>
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csdr flowcontrol <data_rate> <reads_per_second>
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It limits the data rate of a stream to a given `data_rate` number of bytes per second.
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@ -289,7 +289,7 @@ It copies `data_rate / reads_per_second` bytes from the input to the output, doi
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Syntax:
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csdr shift_math_cc <rate>
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csdr shift_math_cc <rate>
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It shifts the signal in the frequency domain by `rate`.
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@ -305,7 +305,7 @@ Internally, a sine and cosine wave is generated, and this function uses `math.h`
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Syntax:
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csdr shift_addition_cc <rate>
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csdr shift_addition_cc <rate>
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Operation is the same as for `shift_math_cc`.
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@ -317,7 +317,7 @@ Internally, this function uses trigonometric addition formulas to generate sine
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Syntax:
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csdr shift_addition_cc_test
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csdr shift_addition_cc_test
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This function was used to test the accuracy of the method above.
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@ -327,7 +327,7 @@ This function was used to test the accuracy of the method above.
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Syntax:
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csdr shift_table_cc <rate> [table_size]
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csdr shift_table_cc <rate> [table_size]
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Operation is the same as with `shift_math_cc`.
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@ -341,7 +341,7 @@ The higher the table size is, the smaller the phase error is.
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Syntax:
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csdr shift_addfast_cc <rate>
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csdr shift_addfast_cc <rate>
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Operation is the same as for `shift_math_cc`.
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@ -353,7 +353,7 @@ Internally, this function uses a NEON-accelerated algorithm on capable systems,
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Syntax:
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csdr shift_unroll_cc <rate>
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csdr shift_unroll_cc <rate>
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Operation is the same as for `shift_math_cc`.
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@ -367,7 +367,7 @@ The loop in this function unrolls quite well if compiled on a PC. It was the fas
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Syntax:
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csdr decimating_shift_addition_cc <rate> [decimation]
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csdr decimating_shift_addition_cc <rate> [decimation]
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It shifts the input signal in the frequency domain, and also decimates it, without filtering. It will be useful as a part of the FFT channelizer implementation (to be done).
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@ -379,7 +379,7 @@ It cannot be used as a channelizer by itself, use `fir_decimate_cc` instead.
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Syntax:
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csdr dcblock_ff
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csdr dcblock_ff
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This is a DC blocking IIR filter.
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@ -389,7 +389,7 @@ This is a DC blocking IIR filter.
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Syntax:
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csdr fastdcblock_ff
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csdr fastdcblock_ff
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This is a DC blocker that works based on the average of the buffer.
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@ -399,7 +399,7 @@ This is a DC blocker that works based on the average of the buffer.
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Syntax:
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csdr fmdemod_atan_cf
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csdr fmdemod_atan_cf
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It is an FM demodulator that internally uses the `atan` function in `math.h`, so it is not so fast.
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@ -409,7 +409,7 @@ It is an FM demodulator that internally uses the `atan` function in `math.h`, so
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Syntax:
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csdr fmdemod_quadri_cf
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csdr fmdemod_quadri_cf
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It is an FM demodulator that is based on the quadri-correlator method, and it can be effectively auto-vectorized, so it should be faster.
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@ -419,7 +419,7 @@ It is an FM demodulator that is based on the quadri-correlator method, and it ca
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Syntax:
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csdr fmdemod_quadri_novect_cf
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csdr fmdemod_quadri_novect_cf
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It has more easily understandable code than the previous one, but can't be auto-vectorized.
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@ -429,7 +429,7 @@ It has more easily understandable code than the previous one, but can't be auto-
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Syntax:
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csdr deemphasis_wfm_ff <sample_rate> <tau>
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csdr deemphasis_wfm_ff <sample_rate> <tau>
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It does de-emphasis with the given RC time constant `tau`.
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@ -443,13 +443,13 @@ In Europe, `tau` should be chosen as `50e-6`, and in the USA, `tau` should be `7
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Syntax:
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csdr deemphasis_nfm_ff <one_of_the_predefined_sample_rates>
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csdr deemphasis_nfm_ff <one_of_the_predefined_sample_rates>
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It does de-emphasis on narrow-band FM for communication equipment (e.g. two-way radios).
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It uses fixed filters so it works only on predefined sample rates, for the actual list of them run:
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cat libcsdr.c | grep DNFMFF_ADD_ARRAY
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cat libcsdr.c | grep DNFMFF_ADD_ARRAY
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----
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@ -457,7 +457,7 @@ It uses fixed filters so it works only on predefined sample rates, for the actua
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Syntax:
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csdr amdemod_cf
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csdr amdemod_cf
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It is an AM demodulator that uses `sqrt`. On some architectures `sqrt` can be directly calculated by dedicated CPU instructions, but on others it may be slower.
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@ -467,7 +467,7 @@ It is an AM demodulator that uses `sqrt`. On some architectures `sqrt` can be di
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Syntax:
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csdr amdemod_estimator_cf
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csdr amdemod_estimator_cf
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It is an AM demodulator that uses an estimation method that is faster but less accurate than `amdemod_cf`.
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@ -477,7 +477,7 @@ It is an AM demodulator that uses an estimation method that is faster but less a
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Syntax:
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csdr firdes_lowpass_f <cutoff_rate> <length> [window [--octave]]
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csdr firdes_lowpass_f <cutoff_rate> <length> [window [--octave]]
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Low-pass FIR filter design function to output real taps, with a `cutoff_rate` proportional to the sampling frequency, using the windowed sinc filter design method.
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@ -501,7 +501,7 @@ The `--octave` parameter lets you directly view the filter response in `octave`.
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Syntax:
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csdr firdes_bandpass_c <low_cut> <high_cut> <length> [window [--octave]]
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csdr firdes_bandpass_c <low_cut> <high_cut> <length> [window [--octave]]
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Band-pass FIR filter design function to output complex taps.
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@ -515,7 +515,7 @@ Other parameters were explained above at `firdes_lowpass_f`.
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Syntax:
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csdr fir_decimate_cc <decimation_factor> [transition_bw [window]]
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csdr fir_decimate_cc <decimation_factor> [transition_bw [window]]
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It is a decimator that keeps one sample out of `decimation_factor` samples.
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|
||||
|
|
@ -527,7 +527,7 @@ To avoid aliasing, it runs a filter on the signal and removes spectral component
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fir_interpolate_cc <interpolation_factor> [transition_bw [window]]
|
||||
csdr fir_interpolate_cc <interpolation_factor> [transition_bw [window]]
|
||||
|
||||
It is an interpolator that generates `interpolation_factor` number of output samples from one input sample.
|
||||
|
||||
|
|
@ -541,7 +541,7 @@ To avoid aliasing, it runs a filter on the signal and removes spectral component
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr rational_resampler_ff <interpolation> <decimation> [transition_bw [window]]
|
||||
csdr rational_resampler_ff <interpolation> <decimation> [transition_bw [window]]
|
||||
|
||||
It is a resampler that takes integer values of `interpolation` and `decimation`.
|
||||
The output sample rate will be `interpolation / decimation × input_sample_rate`.
|
||||
|
|
@ -554,7 +554,7 @@ The output sample rate will be `interpolation / decimation × input_sample_rate`
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fractional_decimator_ff <decimation_rate> [num_poly_points ( [transition_bw [window]] | --prefilter )]
|
||||
csdr fractional_decimator_ff <decimation_rate> [num_poly_points ( [transition_bw [window]] | --prefilter )]
|
||||
|
||||
It can decimate by a floating point ratio.
|
||||
|
||||
|
|
@ -571,7 +571,7 @@ It can filter the signal with an anti-aliasing FIR filter before applying the La
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr old_fractional_decimator_ff <decimation_rate> [transition_bw [window]]
|
||||
csdr old_fractional_decimator_ff <decimation_rate> [transition_bw [window]]
|
||||
|
||||
This is the deprecated, old algorithm to decimate by a floating point ratio, superseded by `fractional_decimator_ff`.
|
||||
|
||||
|
|
@ -583,7 +583,7 @@ This is the deprecated, old algorithm to decimate by a floating point ratio, sup
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr bandpass_fir_fft_cc <low_cut> <high_cut> <transition_bw> [window]
|
||||
csdr bandpass_fir_fft_cc <low_cut> <high_cut> <transition_bw> [window]
|
||||
|
||||
It performs a bandpass FIR filter on complex samples, using FFT and the overlap-add method.
|
||||
|
||||
|
|
@ -595,7 +595,7 @@ Parameters are described under `firdes_bandpass_c` and `firdes_lowpass_f`.
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr agc_ff [hang_time [reference [attack_rate [decay_rate [max_gain [attack_wait [filter_alpha]]]]]]]
|
||||
csdr agc_ff [hang_time [reference [attack_rate [decay_rate [max_gain [attack_wait [filter_alpha]]]]]]]
|
||||
|
||||
It is an automatic gain control function.
|
||||
|
||||
|
|
@ -615,7 +615,7 @@ Its default parameters work best for an audio signal sampled at 48000 Hz.
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fastagc_ff [block_size [reference]]
|
||||
csdr fastagc_ff [block_size [reference]]
|
||||
|
||||
It is a faster AGC that linearly changes the gain, taking the highest amplitude peak in the buffer into consideration. Its output will never exceed `-reference ... reference`.
|
||||
|
||||
|
|
@ -625,7 +625,7 @@ It is a faster AGC that linearly changes the gain, taking the highest amplitude
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fft_cc <fft_size> <out_of_every_n_samples> [window [--octave] [--benchmark]]
|
||||
csdr fft_cc <fft_size> <out_of_every_n_samples> [window [--octave] [--benchmark]]
|
||||
|
||||
It performs an FFT on the first `fft_size` samples out of `out_of_every_n_samples`, thus skipping `out_of_every_n_samples - fft_size` samples in the input.
|
||||
|
||||
|
|
@ -639,7 +639,7 @@ FFTW can be faster if we let it optimalize a while before starting the first tra
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fft_benchmark <fft_size> <fft_cycles> [--benchmark]
|
||||
csdr fft_benchmark <fft_size> <fft_cycles> [--benchmark]
|
||||
|
||||
It measures the time taken to process `fft_cycles` transforms of `fft_size`.
|
||||
It lets FFTW optimalize if used with the `--benchmark` switch.
|
||||
|
|
@ -650,7 +650,7 @@ It lets FFTW optimalize if used with the `--benchmark` switch.
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr logpower_cf [add_db]
|
||||
csdr logpower_cf [add_db]
|
||||
|
||||
Calculates `10*log10(i^2+q^2)+add_db` for the input complex samples. It is useful for drawing power spectrum graphs.
|
||||
|
||||
|
|
@ -666,7 +666,7 @@ Syntax:
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr csdr encode_ima_adpcm_i16_u8
|
||||
csdr csdr encode_ima_adpcm_i16_u8
|
||||
|
||||
Encodes the audio stream to IMA ADPCM, which decreases the size to 25% of the original.
|
||||
|
||||
|
|
@ -676,7 +676,7 @@ Encodes the audio stream to IMA ADPCM, which decreases the size to 25% of the or
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr decode_ima_adpcm_u8_i16
|
||||
csdr decode_ima_adpcm_u8_i16
|
||||
|
||||
Decodes the audio stream from IMA ADPCM.
|
||||
|
||||
|
|
@ -686,7 +686,7 @@ Decodes the audio stream from IMA ADPCM.
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr compress_fft_adpcm_f_u8 <fft_size>
|
||||
csdr compress_fft_adpcm_f_u8 <fft_size>
|
||||
|
||||
Encodes the FFT output vectors of `fft_size`. It should be used on the data output from `logpower_cf`.
|
||||
|
||||
|
|
@ -694,7 +694,7 @@ It resets the ADPCM encoder at the beginning of every vector, and to compensate
|
|||
|
||||
The actual number of padding samples can be determined by running:
|
||||
|
||||
cat csdr.c | grep "define COMPRESS_FFT_PAD_N"
|
||||
cat csdr.c | grep "define COMPRESS_FFT_PAD_N"
|
||||
|
||||
----
|
||||
|
||||
|
|
@ -702,7 +702,7 @@ The actual number of padding samples can be determined by running:
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fft_exchange_sides_ff <fft_size>
|
||||
csdr fft_exchange_sides_ff <fft_size>
|
||||
|
||||
It exchanges the first and second part of the FFT vector, to prepare it for the waterfall/spectrum display. It should operate on the data output from `logpower_cf`.
|
||||
|
||||
|
|
@ -712,7 +712,7 @@ It exchanges the first and second part of the FFT vector, to prepare it for the
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr dsb_fc [q_value]
|
||||
csdr dsb_fc [q_value]
|
||||
|
||||
It converts a real signal to a double sideband complex signal centered around DC.
|
||||
|
||||
|
|
@ -728,7 +728,7 @@ With `q_value = 0` it is an AM-DSB/SC modulator. If you want to get an AM-DSB si
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr add_dcoffset_cc
|
||||
csdr add_dcoffset_cc
|
||||
|
||||
It adds a DC offset to the complex signal: `i_output = 0.5 + i_input / 2, q_output = q_input / 2`
|
||||
|
||||
|
|
@ -738,7 +738,7 @@ It adds a DC offset to the complex signal: `i_output = 0.5 + i_input / 2, q_outp
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr convert_f_samplerf <wait_for_this_sample>
|
||||
csdr convert_f_samplerf <wait_for_this_sample>
|
||||
|
||||
It converts a real signal to the `-mRF` input format of [https://github.com/F5OEO/rpitx](rpitx), so it allows you to generate frequency modulation. The input signal will be the modulating signal. The `<wait_for_this_sample>` parameter is the value for `rpitx` indicating the time to wait between samples. For a sampling rate of 48 ksps, this is 20833.
|
||||
|
||||
|
|
@ -748,7 +748,7 @@ It converts a real signal to the `-mRF` input format of [https://github.com/F5OE
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fmmod_fc
|
||||
csdr fmmod_fc
|
||||
|
||||
It generates a complex FM modulated output from a real input signal.
|
||||
|
||||
|
|
@ -758,7 +758,7 @@ It generates a complex FM modulated output from a real input signal.
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fixed_amplitude_cc <new_amplitude>
|
||||
csdr fixed_amplitude_cc <new_amplitude>
|
||||
|
||||
It changes the amplitude of every complex input sample to a fixed value. It does not change the phase information of the samples.
|
||||
|
||||
|
|
@ -768,17 +768,17 @@ It changes the amplitude of every complex input sample to a fixed value. It does
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr mono2stereo_s16
|
||||
csdr mono2stereo_s16
|
||||
|
||||
It doubles every input sample.
|
||||
|
||||
Example: if the input samples are 16 bit signed integers:
|
||||
|
||||
23 -452 3112
|
||||
23 -452 3112
|
||||
|
||||
The output will be:
|
||||
|
||||
23 23 -452 -452 3112 3112
|
||||
23 23 -452 -452 3112 3112
|
||||
|
||||
----
|
||||
|
||||
|
|
@ -786,11 +786,11 @@ The output will be:
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr setbuf <buffer_size>
|
||||
csdr setbuf <buffer_size>
|
||||
|
||||
See the [buffer sizes](#buffer_sizes) section.
|
||||
|
||||
squelch_and_smeter_cc --fifo <squelch_fifo> --outfifo <smeter_fifo> <use_every_nth> <report_every_nth>
|
||||
squelch_and_smeter_cc --fifo <squelch_fifo> --outfifo <smeter_fifo> <use_every_nth> <report_every_nth>
|
||||
|
||||
This is a controllable squelch, which reads the squelch level input from `<squelch_fifo>` and writes the power level output to `<smeter_fifo>`. Both input and output are in the format of `%g\n`. While calculating the power level, it takes only every `<use_every_nth>` sample into consideration. It writes the S-meter value for every `<report_every_nth>` buffer to `<smeter_fifo>`. If the squelch level is set to 0, it it forces the squelch to be open. If the squelch is closed, it fills the output with zero.
|
||||
|
||||
|
|
@ -800,7 +800,7 @@ This is a controllable squelch, which reads the squelch level input from `<squel
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr fifo <buffer_size> <number_of_buffers>
|
||||
csdr fifo <buffer_size> <number_of_buffers>
|
||||
|
||||
It is similar to `clone`, but internally it uses a circular buffer. It reads as much as possible from the input. It discards input samples if the input buffer is full.
|
||||
|
||||
|
|
@ -810,7 +810,7 @@ It is similar to `clone`, but internally it uses a circular buffer. It reads as
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr psk31_varicode_encoder_u8_u8
|
||||
csdr psk31_varicode_encoder_u8_u8
|
||||
|
||||
It encodes ASCII characters into varicode for PSK31 transmission. It puts a `00` sequence between the varicode characters (which acts as a separator).
|
||||
|
||||
|
|
@ -832,7 +832,7 @@ For this input, the output of `psk31_varicode_encoder_u8_u8` will be the followi
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr repeat_u8 <taps_length> [resonator_rate × N]\n"
|
||||
csdr repeat_u8 <taps_length> [resonator_rate × N]\n"
|
||||
|
||||
It repeatedly outputs a set of data bytes (given with decimal numbers).
|
||||
|
||||
|
|
@ -849,7 +849,7 @@ For example, `csdr repeat_u8 1 1 0 0` will output:
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr uniform_noise_f
|
||||
csdr uniform_noise_f
|
||||
|
||||
It outputs uniform white noise. All samples are within the range [-1.0, 1.0].
|
||||
|
||||
|
|
@ -859,7 +859,7 @@ It outputs uniform white noise. All samples are within the range [-1.0, 1.0].
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr gaussian_noise_c
|
||||
csdr gaussian_noise_c
|
||||
|
||||
It outputs Gaussian white noise. All samples are within the unit circle.
|
||||
|
||||
|
|
@ -869,7 +869,7 @@ It outputs Gaussian white noise. All samples are within the unit circle.
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr pack_bits_8to1_u8_u8
|
||||
csdr pack_bits_8to1_u8_u8
|
||||
|
||||
It serializes the bytes on the input: it outputs each bit of the input byte as a single byte valued 0x00 or 0x01, starting from the lowest bit and going to the highest bit.
|
||||
|
||||
|
|
@ -895,7 +895,7 @@ For consequtive 0x02, 0x03, 0xff bytes on the input, the output will be:
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr awgn_cc <snr_db> [--snrshow]
|
||||
csdr awgn_cc <snr_db> [--snrshow]
|
||||
|
||||
It adds white noise with the given SNR to a signal assumed to be of 0 dB power.
|
||||
|
||||
|
|
@ -907,7 +907,7 @@ If the `--snrshow` switch is given, it also shows the actual SNR based on the ca
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr add_n_zero_samples_at_beginning_f <n_zero_samples>
|
||||
csdr add_n_zero_samples_at_beginning_f <n_zero_samples>
|
||||
|
||||
When the function is executed, it furst writes `<n_zero_samples>` 32-bit floating point zeros at the output, after that it just clones the input at the output.
|
||||
|
||||
|
|
@ -917,7 +917,7 @@ When the function is executed, it furst writes `<n_zero_samples>` 32-bit floatin
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr ?<search_the_function_list>
|
||||
csdr ?<search_the_function_list>
|
||||
|
||||
You can search the functions available in `csdr` just as if you typed: `csdr 2>&1 | grep <search_what>`
|
||||
|
||||
|
|
@ -925,7 +925,7 @@ You can search the functions available in `csdr` just as if you typed: `csdr 2>&
|
|||
|
||||
Syntax:
|
||||
|
||||
csdr =<evaluate_python_expression>
|
||||
csdr =<evaluate_python_expression>
|
||||
|
||||
When running complicated `csdr` commands, we usually run into using `python` to calculate certain parameters.
|
||||
|
||||
|
|
@ -933,19 +933,19 @@ This function can eliminate some typing and make our command clearer.
|
|||
|
||||
Instead of having to write:
|
||||
|
||||
csdr shift_addition_cc $(python -c "print 1200/2400000.")
|
||||
csdr shift_addition_cc $(python -c "print 1200/2400000.")
|
||||
|
||||
...we can type:
|
||||
|
||||
csdr shift_addition_cc $(csdr =1200/2400000.)
|
||||
csdr shift_addition_cc $(csdr =1200/2400000.)
|
||||
|
||||
If using parenthesis inside the expression, it needs to be escaped (as `bash` would want to parse it):
|
||||
|
||||
csdr shift_addition_cc $(csdr =\(1200+300\)/2400000)
|
||||
csdr shift_addition_cc $(csdr =\(1200+300\)/2400000)
|
||||
|
||||
Another solution is using single quotes to wrap the expression:
|
||||
|
||||
csdr shift_addition_cc $(csdr '=(1200+300)/2400000.')
|
||||
csdr shift_addition_cc $(csdr '=(1200+300)/2400000.')
|
||||
|
||||
Current version of `csdr` executes the following python script for this function:
|
||||
|
||||
|
|
@ -961,21 +961,21 @@ This means that one can also call math functions like `sqrt()`.
|
|||
|
||||
Some parameters can be changed while the `csdr` process is running. To achieve this, some `csdr` functions have special parameters. You have to supply a fifo previously created by the `mkfifo` command. Processing will only start after the first control command has been received by `csdr` over the FIFO.
|
||||
|
||||
shift_addition_cc --fifo <fifo_path>
|
||||
shift_addition_cc --fifo <fifo_path>
|
||||
|
||||
By writing to the given FIFO file with the syntax below, you can control the shift rate:
|
||||
|
||||
<shift_rate>\n
|
||||
<shift_rate>\n
|
||||
|
||||
E.g. you can send `-0.3\n`
|
||||
|
||||
Processing will only start after the first control command has been received by `csdr` over the FIFO.
|
||||
|
||||
bandpass_fir_fft_cc --fifo <fifo_path> <transition_bw> [window]
|
||||
bandpass_fir_fft_cc --fifo <fifo_path> <transition_bw> [window]
|
||||
|
||||
By writing to the given FIFO file with the syntax below, you can control the shift rate:
|
||||
|
||||
<low_cut> <high_cut>\n
|
||||
<low_cut> <high_cut>\n
|
||||
|
||||
E.g. you can send `-0.05 0.02\n`
|
||||
|
||||
|
|
@ -1043,17 +1043,17 @@ To compile *sdr.js*, first get <a href="http://emscripten.org/">Emscripten</a>.
|
|||
|
||||
To install and build dependencies (for now, only FFTW3):
|
||||
|
||||
make emcc-get-deps
|
||||
make emcc-get-deps
|
||||
|
||||
To compile *sdr.js* (which will be created in the `sdr.js` subdirectory):
|
||||
|
||||
make emcc
|
||||
make emcc
|
||||
|
||||
You can test *sdr.js* by opening *sdr.html*. It contains a test for *firdes_lowpass_f* for this time.
|
||||
|
||||
To remove *sdr.js* and the compiled dependencies:
|
||||
|
||||
make emcc-clean
|
||||
make emcc-clean
|
||||
|
||||
## [nmux](#nmux)
|
||||
|
||||
|
|
|
|||
180
libcsdr.h
180
libcsdr.h
|
|
@ -69,7 +69,7 @@ typedef struct complexf_s { float i; float q; } complexf;
|
|||
//window
|
||||
typedef enum window_s
|
||||
{
|
||||
WINDOW_BOXCAR, WINDOW_BLACKMAN, WINDOW_HAMMING
|
||||
WINDOW_BOXCAR, WINDOW_BLACKMAN, WINDOW_HAMMING
|
||||
} window_t;
|
||||
|
||||
#define WINDOW_DEFAULT WINDOW_HAMMING
|
||||
|
|
@ -109,31 +109,31 @@ float shift_math_cc(complexf *input, complexf* output, int input_size, float rat
|
|||
|
||||
typedef struct dcblock_preserve_s
|
||||
{
|
||||
float last_input;
|
||||
float last_output;
|
||||
float last_input;
|
||||
float last_output;
|
||||
} dcblock_preserve_t;
|
||||
dcblock_preserve_t dcblock_ff(float* input, float* output, int input_size, float a, dcblock_preserve_t preserved);
|
||||
float fastdcblock_ff(float* input, float* output, int input_size, float last_dc_level);
|
||||
|
||||
typedef struct fastagc_ff_s
|
||||
{
|
||||
float* buffer_1;
|
||||
float* buffer_2;
|
||||
float* buffer_input; //it is the actual input buffer to fill
|
||||
float peak_1;
|
||||
float peak_2;
|
||||
int input_size;
|
||||
float reference;
|
||||
float last_gain;
|
||||
float* buffer_1;
|
||||
float* buffer_2;
|
||||
float* buffer_input; //it is the actual input buffer to fill
|
||||
float peak_1;
|
||||
float peak_2;
|
||||
int input_size;
|
||||
float reference;
|
||||
float last_gain;
|
||||
} fastagc_ff_t;
|
||||
|
||||
void fastagc_ff(fastagc_ff_t* input, float* output);
|
||||
|
||||
typedef struct rational_resampler_ff_s
|
||||
{
|
||||
int input_processed;
|
||||
int output_size;
|
||||
int last_taps_delay;
|
||||
int input_processed;
|
||||
int output_size;
|
||||
int last_taps_delay;
|
||||
} rational_resampler_ff_t;
|
||||
|
||||
rational_resampler_ff_t rational_resampler_ff(float *input, float *output, int input_size, int interpolation, int decimation, float *taps, int taps_length, int last_taps_delay);
|
||||
|
|
@ -149,37 +149,37 @@ void log_ff(float* input, float* output, int size, float add_db);
|
|||
|
||||
typedef struct fractional_decimator_ff_s
|
||||
{
|
||||
float where;
|
||||
int input_processed;
|
||||
int output_size;
|
||||
int num_poly_points; //number of samples that the Lagrange interpolator will use
|
||||
float* poly_precalc_denomiator; //while we don't precalculate coefficients here as in a Farrow structure, because it is a fractional interpolator, but we rather precaculate part of the interpolator expression
|
||||
//float* last_inputs_circbuf; //circular buffer to store the last (num_poly_points) number of input samples.
|
||||
//int last_inputs_startsat; //where the circular buffer starts now
|
||||
//int last_inputs_samplewhere;
|
||||
float* coeffs_buf;
|
||||
float* filtered_buf;
|
||||
int xifirst;
|
||||
int xilast;
|
||||
float rate;
|
||||
float *taps;
|
||||
int taps_length;
|
||||
float where;
|
||||
int input_processed;
|
||||
int output_size;
|
||||
int num_poly_points; //number of samples that the Lagrange interpolator will use
|
||||
float* poly_precalc_denomiator; //while we don't precalculate coefficients here as in a Farrow structure, because it is a fractional interpolator, but we rather precaculate part of the interpolator expression
|
||||
//float* last_inputs_circbuf; //circular buffer to store the last (num_poly_points) number of input samples.
|
||||
//int last_inputs_startsat; //where the circular buffer starts now
|
||||
//int last_inputs_samplewhere;
|
||||
float* coeffs_buf;
|
||||
float* filtered_buf;
|
||||
int xifirst;
|
||||
int xilast;
|
||||
float rate;
|
||||
float *taps;
|
||||
int taps_length;
|
||||
} fractional_decimator_ff_t;
|
||||
fractional_decimator_ff_t fractional_decimator_ff_init(float rate, int num_poly_points, float* taps, int taps_length);
|
||||
void fractional_decimator_ff(float* input, float* output, int input_size, fractional_decimator_ff_t* d);
|
||||
|
||||
typedef struct old_fractional_decimator_ff_s
|
||||
{
|
||||
float remain;
|
||||
int input_processed;
|
||||
int output_size;
|
||||
float remain;
|
||||
int input_processed;
|
||||
int output_size;
|
||||
} old_fractional_decimator_ff_t;
|
||||
old_fractional_decimator_ff_t old_fractional_decimator_ff(float* input, float* output, int input_size, float rate, float *taps, int taps_length, old_fractional_decimator_ff_t d);
|
||||
|
||||
typedef struct shift_table_data_s
|
||||
{
|
||||
float* table;
|
||||
int table_size;
|
||||
float* table;
|
||||
int table_size;
|
||||
} shift_table_data_t;
|
||||
void shift_table_deinit(shift_table_data_t table_data);
|
||||
shift_table_data_t shift_table_init(int table_size);
|
||||
|
|
@ -187,9 +187,9 @@ float shift_table_cc(complexf* input, complexf* output, int input_size, float ra
|
|||
|
||||
typedef struct shift_addfast_data_s
|
||||
{
|
||||
float dsin[4];
|
||||
float dcos[4];
|
||||
float phase_increment;
|
||||
float dsin[4];
|
||||
float dcos[4];
|
||||
float phase_increment;
|
||||
} shift_addfast_data_t;
|
||||
shift_addfast_data_t shift_addfast_init(float rate);
|
||||
shift_addfast_data_t shift_addfast_init(float rate);
|
||||
|
|
@ -197,10 +197,10 @@ float shift_addfast_cc(complexf *input, complexf* output, int input_size, shift_
|
|||
|
||||
typedef struct shift_unroll_data_s
|
||||
{
|
||||
float* dsin;
|
||||
float* dcos;
|
||||
float phase_increment;
|
||||
int size;
|
||||
float* dsin;
|
||||
float* dcos;
|
||||
float phase_increment;
|
||||
int size;
|
||||
} shift_unroll_data_t;
|
||||
float shift_unroll_cc(complexf *input, complexf* output, int input_size, shift_unroll_data_t* d, float starting_phase);
|
||||
shift_unroll_data_t shift_unroll_init(float rate, int size);
|
||||
|
|
@ -234,25 +234,25 @@ int is_nan(float f);
|
|||
|
||||
typedef struct rtty_baudot_item_s
|
||||
{
|
||||
unsigned long long code;
|
||||
unsigned char ascii_letter;
|
||||
unsigned char ascii_figure;
|
||||
unsigned long long code;
|
||||
unsigned char ascii_letter;
|
||||
unsigned char ascii_figure;
|
||||
} rtty_baudot_item_t;
|
||||
|
||||
typedef enum rtty_baudot_decoder_state_e
|
||||
{
|
||||
RTTY_BAUDOT_WAITING_STOP_PULSE = 0,
|
||||
RTTY_BAUDOT_WAITING_START_PULSE,
|
||||
RTTY_BAUDOT_RECEIVING_DATA
|
||||
RTTY_BAUDOT_WAITING_STOP_PULSE = 0,
|
||||
RTTY_BAUDOT_WAITING_START_PULSE,
|
||||
RTTY_BAUDOT_RECEIVING_DATA
|
||||
} rtty_baudot_decoder_state_t;
|
||||
|
||||
typedef struct rtty_baudot_decoder_s
|
||||
{
|
||||
unsigned char fig_mode;
|
||||
unsigned char character_received;
|
||||
unsigned short shr;
|
||||
unsigned char bit_cntr;
|
||||
rtty_baudot_decoder_state_t state;
|
||||
unsigned char fig_mode;
|
||||
unsigned char character_received;
|
||||
unsigned short shr;
|
||||
unsigned char bit_cntr;
|
||||
rtty_baudot_decoder_state_t state;
|
||||
} rtty_baudot_decoder_t;
|
||||
|
||||
#define RTTY_FIGURE_MODE_SELECT_CODE 0b11011
|
||||
|
|
@ -265,9 +265,9 @@ char rtty_baudot_decoder_push(rtty_baudot_decoder_t* s, unsigned char symbol);
|
|||
|
||||
typedef struct psk31_varicode_item_s
|
||||
{
|
||||
unsigned long long code;
|
||||
int bitcount;
|
||||
unsigned char ascii;
|
||||
unsigned long long code;
|
||||
int bitcount;
|
||||
unsigned char ascii;
|
||||
} psk31_varicode_item_t;
|
||||
|
||||
char psk31_varicode_decoder_push(unsigned long long* status_shr, unsigned char symbol);
|
||||
|
|
@ -276,12 +276,12 @@ char psk31_varicode_decoder_push(unsigned long long* status_shr, unsigned char s
|
|||
|
||||
typedef struct serial_line_s
|
||||
{
|
||||
float samples_per_bits;
|
||||
int databits; //including parity
|
||||
float stopbits;
|
||||
int output_size;
|
||||
int input_used;
|
||||
float bit_sampling_width_ratio;
|
||||
float samples_per_bits;
|
||||
int databits; //including parity
|
||||
float stopbits;
|
||||
int output_size;
|
||||
int input_used;
|
||||
float bit_sampling_width_ratio;
|
||||
} serial_line_t;
|
||||
|
||||
void serial_line_decoder_f_u8(serial_line_t* s, float* input, unsigned char* output, int input_size);
|
||||
|
|
@ -290,20 +290,20 @@ void binary_slicer_f_u8(float* input, unsigned char* output, int input_size);
|
|||
|
||||
typedef enum pll_type_e
|
||||
{
|
||||
PLL_P_CONTROLLER=1,
|
||||
PLL_PI_CONTROLLER=2
|
||||
PLL_P_CONTROLLER=1,
|
||||
PLL_PI_CONTROLLER=2
|
||||
} pll_type_t;
|
||||
|
||||
typedef struct pll_s
|
||||
{
|
||||
pll_type_t pll_type;
|
||||
//common:
|
||||
float output_phase;
|
||||
float dphase;
|
||||
float frequency;
|
||||
float alpha;
|
||||
float beta;
|
||||
float iir_temp;
|
||||
pll_type_t pll_type;
|
||||
//common:
|
||||
float output_phase;
|
||||
float dphase;
|
||||
float frequency;
|
||||
float alpha;
|
||||
float beta;
|
||||
float iir_temp;
|
||||
} pll_t;
|
||||
|
||||
void pll_cc_init_pi_controller(pll_t* p, float bandwidth, float ko, float kd, float damping_factor);
|
||||
|
|
@ -312,25 +312,25 @@ void pll_cc(pll_t* p, complexf* input, float* output_dphase, complexf* output_nc
|
|||
|
||||
typedef enum timing_recovery_algorithm_e
|
||||
{
|
||||
TIMING_RECOVERY_ALGORITHM_GARDNER,
|
||||
TIMING_RECOVERY_ALGORITHM_EARLYLATE
|
||||
TIMING_RECOVERY_ALGORITHM_GARDNER,
|
||||
TIMING_RECOVERY_ALGORITHM_EARLYLATE
|
||||
} timing_recovery_algorithm_t;
|
||||
|
||||
#define TIMING_RECOVERY_ALGORITHM_DEFAULT TIMING_RECOVERY_ALGORITHM_GARDNER
|
||||
|
||||
typedef struct timing_recovery_state_s
|
||||
{
|
||||
timing_recovery_algorithm_t algorithm;
|
||||
int decimation_rate; // = input_rate / output_rate. We should get an input signal that is N times oversampled.
|
||||
int output_size;
|
||||
int input_processed;
|
||||
int use_q; //use both I and Q for calculating the error
|
||||
int debug_phase;
|
||||
int debug_count;
|
||||
int debug_force;
|
||||
int debug_writefiles;
|
||||
int last_correction_offset;
|
||||
float earlylate_ratio;
|
||||
timing_recovery_algorithm_t algorithm;
|
||||
int decimation_rate; // = input_rate / output_rate. We should get an input signal that is N times oversampled.
|
||||
int output_size;
|
||||
int input_processed;
|
||||
int use_q; //use both I and Q for calculating the error
|
||||
int debug_phase;
|
||||
int debug_count;
|
||||
int debug_force;
|
||||
int debug_writefiles;
|
||||
int last_correction_offset;
|
||||
float earlylate_ratio;
|
||||
} timing_recovery_state_t;
|
||||
|
||||
timing_recovery_state_t timing_recovery_init(timing_recovery_algorithm_t algorithm, int decimation_rate, int use_q);
|
||||
|
|
@ -348,12 +348,12 @@ unsigned char differential_codec(unsigned char* input, unsigned char* output, in
|
|||
|
||||
typedef struct bpsk_costas_loop_state_s
|
||||
{
|
||||
float rc_filter_alpha;
|
||||
float vco_phase_addition_multiplier;
|
||||
float vco_phase;
|
||||
float last_lpfi_output;
|
||||
float last_lpfq_output;
|
||||
float last_vco_phase_addition;
|
||||
float rc_filter_alpha;
|
||||
float vco_phase_addition_multiplier;
|
||||
float vco_phase;
|
||||
float last_lpfi_output;
|
||||
float last_lpfq_output;
|
||||
float last_vco_phase_addition;
|
||||
} bpsk_costas_loop_state_t;
|
||||
|
||||
bpsk_costas_loop_state_t init_bpsk_costas_loop_cc(float samples_per_bits);
|
||||
|
|
|
|||
Ładowanie…
Reference in New Issue