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@ -13,8 +13,8 @@ Most of the code is available under the permissive BSD license, with some option
How to compile How to compile
-------------- --------------
make make
sudo make install sudo make install
The project was only tested on Linux. It has the following dependencies: `libfftw3-dev` The project was only tested on Linux. It has the following dependencies: `libfftw3-dev`
@ -24,7 +24,7 @@ To run the examples, you will also need <a href="http://sdr.osmocom.org/trac/wik
If you compile *fftw3* from sources for use with *libcsdr*, you need to configure it with 32-bit float support enabled: If you compile *fftw3* from sources for use with *libcsdr*, you need to configure it with 32-bit float support enabled:
./configure --enable-float ./configure --enable-float
(This is for *fftw3*, not *libcsdr*. You do not need to run the configure script before compiling *libcsdr*.) (This is for *fftw3*, not *libcsdr*. You do not need to run the configure script before compiling *libcsdr*.)
@ -39,7 +39,7 @@ Usage by example
### Demodulate WFM ### Demodulate WFM
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 - 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 -
- 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. - 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.
- 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`. - 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`.
@ -51,7 +51,7 @@ Usage by example
### Demodulate WFM: advanced ### Demodulate WFM: advanced
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 - 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 -
- 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. - 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.
- 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. - 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.
@ -60,19 +60,19 @@ Usage by example
Sample rates look like this: Sample rates look like this:
2.4 Msps 240 ksps 48 ksps 2.4 Msps 240 ksps 48 ksps
I/Q source ------------> FIR decimation ------------> FM demod -> frac. decimation ---------> deemphasis -> sound card I/Q source ------------> FIR decimation ------------> FM demod -> frac. decimation ---------> deemphasis -> sound card
*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: *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:
csdr-fm 89.5 20 csdr-fm 89.5 20
The first parameter is the frequency in MHz, and the second optional parameter is the RTL-SDR tuner gain in dB. The first parameter is the frequency in MHz, and the second optional parameter is the RTL-SDR tuner gain in dB.
### Demodulate NFM ### Demodulate NFM
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 - 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 -
- Note that the decimation factor is higher (we want to select a ~25 kHz channel). - Note that the decimation factor is higher (we want to select a ~25 kHz channel).
- Also there is a python hack to calculate the relative shift offset. The real receiver frequency is `145350000` Hz. - Also there is a python hack to calculate the relative shift offset. The real receiver frequency is `145350000` Hz.
@ -80,14 +80,14 @@ The first parameter is the frequency in MHz, and the second optional parameter i
### Demodulate AM ### Demodulate AM
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 - 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 -
- `amdemod_cf` is used as demodulator. - `amdemod_cf` is used as demodulator.
- `agc_ff` should be used for AM and SSB. - `agc_ff` should be used for AM and SSB.
### Design FIR band-pass filter (with complex taps) ### Design FIR band-pass filter (with complex taps)
csdr firdes_bandpass_c 0 0.5 59 HAMMING --octave | octave -i csdr firdes_bandpass_c 0 0.5 59 HAMMING --octave | octave -i
- ...and then plot its frequency response with octave. (You can close octave window by issuing Ctrl-C in the terminal window.) - ...and then plot its frequency response with octave. (You can close octave window by issuing Ctrl-C in the terminal window.)
- 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). - 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).
@ -95,13 +95,13 @@ The first parameter is the frequency in MHz, and the second optional parameter i
### Demodulate SSB ### Demodulate SSB
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 - 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 -
- 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`. - 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`.
### Draw FFT ### Draw FFT
rtl_sdr -s 2400000 -f 104300000 -g 20 - | csdr convert_u8_f | csdr fft_cc 1024 1200000 HAMMING --octave | octave -i > /dev/null rtl_sdr -s 2400000 -f 104300000 -g 20 - | csdr convert_u8_f | csdr fft_cc 1024 1200000 HAMMING --octave | octave -i > /dev/null
- 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.) - 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.)
- 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. - 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.
@ -153,7 +153,7 @@ Optional parameters have safe defaults, for more info look at the code.
Syntax: Syntax:
csdr realpart_cf csdr realpart_cf
It takes the real part of the complex signal, and throws away the imaginary part. It takes the real part of the complex signal, and throws away the imaginary part.
@ -163,7 +163,7 @@ It takes the real part of the complex signal, and throws away the imaginary part
Syntax: Syntax:
csdr clipdetect_ff csdr clipdetect_ff
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. 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.
@ -173,7 +173,7 @@ It clones the signal (the input and the output is the same), but it prints a war
Syntax: Syntax:
csdr limit_ff [max_amplitude] csdr limit_ff [max_amplitude]
The input signal amplitude will not be let out of the `-max_amplitude ... max_amplitude` range. The input signal amplitude will not be let out of the `-max_amplitude ... max_amplitude` range.
@ -183,7 +183,7 @@ The input signal amplitude will not be let out of the `-max_amplitude ... max_am
Syntax: Syntax:
csdr gain_ff <gain> csdr gain_ff <gain>
It multiplies all samples by `gain`. It multiplies all samples by `gain`.
@ -193,7 +193,7 @@ It multiplies all samples by `gain`.
Syntax: Syntax:
csdr clone csdr clone
It copies the input to the output. It copies the input to the output.
@ -203,7 +203,7 @@ It copies the input to the output.
Syntax: Syntax:
csdr through csdr through
It copies the input to the output, while also displaying the data rate going through it. It copies the input to the output, while also displaying the data rate going through it.
@ -213,7 +213,7 @@ It copies the input to the output, while also displaying the data rate going thr
Syntax: Syntax:
csdr none csdr none
The `csdr` process just exits with 0. The `csdr` process just exits with 0.
@ -223,7 +223,7 @@ The `csdr` process just exits with 0.
Syntax: Syntax:
csdr yes_f <to_repeat> [buf_times] csdr yes_f <to_repeat> [buf_times]
It outputs continously the `to_repeat` float number. It outputs continously the `to_repeat` float number.
@ -237,7 +237,7 @@ Else, after outputing `buf_times` number of buffers (the size of which is stated
Syntax: Syntax:
csdr detect_nan_ff csdr detect_nan_ff
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. 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.
@ -247,7 +247,7 @@ Along with copying its input samples to the output, it prints a warning message
Syntax: Syntax:
csdr dump_f csdr dump_f
It prints all floating point input samples as text. It prints all floating point input samples as text.
@ -265,7 +265,7 @@ Aliases for this function: `floatdump_f`
Syntax: Syntax:
csdr dump_u8 csdr dump_u8
It prints all input bytes as text, in hexadecimal form. It prints all input bytes as text, in hexadecimal form.
@ -277,7 +277,7 @@ Alternatively, you can use the `xxd` command (built into most Linux distribution
Syntax: Syntax:
csdr flowcontrol <data_rate> <reads_per_second> csdr flowcontrol <data_rate> <reads_per_second>
It limits the data rate of a stream to a given `data_rate` number of bytes per second. It limits the data rate of a stream to a given `data_rate` number of bytes per second.
@ -289,7 +289,7 @@ It copies `data_rate / reads_per_second` bytes from the input to the output, doi
Syntax: Syntax:
csdr shift_math_cc <rate> csdr shift_math_cc <rate>
It shifts the signal in the frequency domain by `rate`. It shifts the signal in the frequency domain by `rate`.
@ -305,7 +305,7 @@ Internally, a sine and cosine wave is generated, and this function uses `math.h`
Syntax: Syntax:
csdr shift_addition_cc <rate> csdr shift_addition_cc <rate>
Operation is the same as for `shift_math_cc`. Operation is the same as for `shift_math_cc`.
@ -317,7 +317,7 @@ Internally, this function uses trigonometric addition formulas to generate sine
Syntax: Syntax:
csdr shift_addition_cc_test csdr shift_addition_cc_test
This function was used to test the accuracy of the method above. This function was used to test the accuracy of the method above.
@ -327,7 +327,7 @@ This function was used to test the accuracy of the method above.
Syntax: Syntax:
csdr shift_table_cc <rate> [table_size] csdr shift_table_cc <rate> [table_size]
Operation is the same as with `shift_math_cc`. Operation is the same as with `shift_math_cc`.
@ -341,7 +341,7 @@ The higher the table size is, the smaller the phase error is.
Syntax: Syntax:
csdr shift_addfast_cc <rate> csdr shift_addfast_cc <rate>
Operation is the same as for `shift_math_cc`. Operation is the same as for `shift_math_cc`.
@ -353,7 +353,7 @@ Internally, this function uses a NEON-accelerated algorithm on capable systems,
Syntax: Syntax:
csdr shift_unroll_cc <rate> csdr shift_unroll_cc <rate>
Operation is the same as for `shift_math_cc`. Operation is the same as for `shift_math_cc`.
@ -367,7 +367,7 @@ The loop in this function unrolls quite well if compiled on a PC. It was the fas
Syntax: Syntax:
csdr decimating_shift_addition_cc <rate> [decimation] csdr decimating_shift_addition_cc <rate> [decimation]
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). 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).
@ -379,7 +379,7 @@ It cannot be used as a channelizer by itself, use `fir_decimate_cc` instead.
Syntax: Syntax:
csdr dcblock_ff csdr dcblock_ff
This is a DC blocking IIR filter. This is a DC blocking IIR filter.
@ -389,7 +389,7 @@ This is a DC blocking IIR filter.
Syntax: Syntax:
csdr fastdcblock_ff csdr fastdcblock_ff
This is a DC blocker that works based on the average of the buffer. This is a DC blocker that works based on the average of the buffer.
@ -399,7 +399,7 @@ This is a DC blocker that works based on the average of the buffer.
Syntax: Syntax:
csdr fmdemod_atan_cf csdr fmdemod_atan_cf
It is an FM demodulator that internally uses the `atan` function in `math.h`, so it is not so fast. It is an FM demodulator that internally uses the `atan` function in `math.h`, so it is not so fast.
@ -409,7 +409,7 @@ It is an FM demodulator that internally uses the `atan` function in `math.h`, so
Syntax: Syntax:
csdr fmdemod_quadri_cf csdr fmdemod_quadri_cf
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. 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.
@ -419,7 +419,7 @@ It is an FM demodulator that is based on the quadri-correlator method, and it ca
Syntax: Syntax:
csdr fmdemod_quadri_novect_cf csdr fmdemod_quadri_novect_cf
It has more easily understandable code than the previous one, but can't be auto-vectorized. It has more easily understandable code than the previous one, but can't be auto-vectorized.
@ -429,7 +429,7 @@ It has more easily understandable code than the previous one, but can't be auto-
Syntax: Syntax:
csdr deemphasis_wfm_ff <sample_rate> <tau> csdr deemphasis_wfm_ff <sample_rate> <tau>
It does de-emphasis with the given RC time constant `tau`. It does de-emphasis with the given RC time constant `tau`.
@ -443,13 +443,13 @@ In Europe, `tau` should be chosen as `50e-6`, and in the USA, `tau` should be `7
Syntax: Syntax:
csdr deemphasis_nfm_ff <one_of_the_predefined_sample_rates> csdr deemphasis_nfm_ff <one_of_the_predefined_sample_rates>
It does de-emphasis on narrow-band FM for communication equipment (e.g. two-way radios). It does de-emphasis on narrow-band FM for communication equipment (e.g. two-way radios).
It uses fixed filters so it works only on predefined sample rates, for the actual list of them run: It uses fixed filters so it works only on predefined sample rates, for the actual list of them run:
cat libcsdr.c | grep DNFMFF_ADD_ARRAY cat libcsdr.c | grep DNFMFF_ADD_ARRAY
---- ----
@ -457,7 +457,7 @@ It uses fixed filters so it works only on predefined sample rates, for the actua
Syntax: Syntax:
csdr amdemod_cf csdr amdemod_cf
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. 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.
@ -467,7 +467,7 @@ It is an AM demodulator that uses `sqrt`. On some architectures `sqrt` can be di
Syntax: Syntax:
csdr amdemod_estimator_cf csdr amdemod_estimator_cf
It is an AM demodulator that uses an estimation method that is faster but less accurate than `amdemod_cf`. It is an AM demodulator that uses an estimation method that is faster but less accurate than `amdemod_cf`.
@ -477,7 +477,7 @@ It is an AM demodulator that uses an estimation method that is faster but less a
Syntax: Syntax:
csdr firdes_lowpass_f <cutoff_rate> <length> [window [--octave]] csdr firdes_lowpass_f <cutoff_rate> <length> [window [--octave]]
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. 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.
@ -501,7 +501,7 @@ The `--octave` parameter lets you directly view the filter response in `octave`.
Syntax: Syntax:
csdr firdes_bandpass_c <low_cut> <high_cut> <length> [window [--octave]] csdr firdes_bandpass_c <low_cut> <high_cut> <length> [window [--octave]]
Band-pass FIR filter design function to output complex taps. Band-pass FIR filter design function to output complex taps.
@ -515,7 +515,7 @@ Other parameters were explained above at `firdes_lowpass_f`.
Syntax: Syntax:
csdr fir_decimate_cc <decimation_factor> [transition_bw [window]] csdr fir_decimate_cc <decimation_factor> [transition_bw [window]]
It is a decimator that keeps one sample out of `decimation_factor` samples. It is a decimator that keeps one sample out of `decimation_factor` samples.
@ -527,7 +527,7 @@ To avoid aliasing, it runs a filter on the signal and removes spectral component
Syntax: 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. 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: 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`. It is a resampler that takes integer values of `interpolation` and `decimation`.
The output sample rate will be `interpolation / decimation × input_sample_rate`. 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: 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. 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: 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`. 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: 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. 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: 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. 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: 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`. 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: 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. 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: 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 measures the time taken to process `fft_cycles` transforms of `fft_size`.
It lets FFTW optimalize if used with the `--benchmark` switch. 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: 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. 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: 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. 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: Syntax:
csdr decode_ima_adpcm_u8_i16 csdr decode_ima_adpcm_u8_i16
Decodes the audio stream from IMA ADPCM. Decodes the audio stream from IMA ADPCM.
@ -686,7 +686,7 @@ Decodes the audio stream from IMA ADPCM.
Syntax: 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`. 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: 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: 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`. 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: 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. 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: 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` 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: 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. 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: Syntax:
csdr fmmod_fc csdr fmmod_fc
It generates a complex FM modulated output from a real input signal. 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: 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. 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: Syntax:
csdr mono2stereo_s16 csdr mono2stereo_s16
It doubles every input sample. It doubles every input sample.
Example: if the input samples are 16 bit signed integers: Example: if the input samples are 16 bit signed integers:
23 -452 3112 23 -452 3112
The output will be: 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: Syntax:
csdr setbuf <buffer_size> csdr setbuf <buffer_size>
See the [buffer sizes](#buffer_sizes) section. 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. 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: 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. 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: 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). 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: 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). 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: Syntax:
csdr uniform_noise_f csdr uniform_noise_f
It outputs uniform white noise. All samples are within the range [-1.0, 1.0]. 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: Syntax:
csdr gaussian_noise_c csdr gaussian_noise_c
It outputs Gaussian white noise. All samples are within the unit circle. 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: 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. 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: 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. 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: 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. 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: 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>` 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: Syntax:
csdr =<evaluate_python_expression> csdr =<evaluate_python_expression>
When running complicated `csdr` commands, we usually run into using `python` to calculate certain parameters. 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: 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: ...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): 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: 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: 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. 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: 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` 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. 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: 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` 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): 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): 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. 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: To remove *sdr.js* and the compiled dependencies:
make emcc-clean make emcc-clean
## [nmux](#nmux) ## [nmux](#nmux)

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@ -69,7 +69,7 @@ typedef struct complexf_s { float i; float q; } complexf;
//window //window
typedef enum window_s typedef enum window_s
{ {
WINDOW_BOXCAR, WINDOW_BLACKMAN, WINDOW_HAMMING WINDOW_BOXCAR, WINDOW_BLACKMAN, WINDOW_HAMMING
} window_t; } window_t;
#define WINDOW_DEFAULT WINDOW_HAMMING #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 typedef struct dcblock_preserve_s
{ {
float last_input; float last_input;
float last_output; float last_output;
} dcblock_preserve_t; } dcblock_preserve_t;
dcblock_preserve_t dcblock_ff(float* input, float* output, int input_size, float a, dcblock_preserve_t preserved); 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); float fastdcblock_ff(float* input, float* output, int input_size, float last_dc_level);
typedef struct fastagc_ff_s typedef struct fastagc_ff_s
{ {
float* buffer_1; float* buffer_1;
float* buffer_2; float* buffer_2;
float* buffer_input; //it is the actual input buffer to fill float* buffer_input; //it is the actual input buffer to fill
float peak_1; float peak_1;
float peak_2; float peak_2;
int input_size; int input_size;
float reference; float reference;
float last_gain; float last_gain;
} fastagc_ff_t; } fastagc_ff_t;
void fastagc_ff(fastagc_ff_t* input, float* output); void fastagc_ff(fastagc_ff_t* input, float* output);
typedef struct rational_resampler_ff_s typedef struct rational_resampler_ff_s
{ {
int input_processed; int input_processed;
int output_size; int output_size;
int last_taps_delay; int last_taps_delay;
} rational_resampler_ff_t; } 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); 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 typedef struct fractional_decimator_ff_s
{ {
float where; float where;
int input_processed; int input_processed;
int output_size; int output_size;
int num_poly_points; //number of samples that the Lagrange interpolator will use 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* 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. //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_startsat; //where the circular buffer starts now
//int last_inputs_samplewhere; //int last_inputs_samplewhere;
float* coeffs_buf; float* coeffs_buf;
float* filtered_buf; float* filtered_buf;
int xifirst; int xifirst;
int xilast; int xilast;
float rate; float rate;
float *taps; float *taps;
int taps_length; int taps_length;
} fractional_decimator_ff_t; } fractional_decimator_ff_t;
fractional_decimator_ff_t fractional_decimator_ff_init(float rate, int num_poly_points, float* taps, int taps_length); 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); void fractional_decimator_ff(float* input, float* output, int input_size, fractional_decimator_ff_t* d);
typedef struct old_fractional_decimator_ff_s typedef struct old_fractional_decimator_ff_s
{ {
float remain; float remain;
int input_processed; int input_processed;
int output_size; int output_size;
} old_fractional_decimator_ff_t; } 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); 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 typedef struct shift_table_data_s
{ {
float* table; float* table;
int table_size; int table_size;
} shift_table_data_t; } shift_table_data_t;
void shift_table_deinit(shift_table_data_t table_data); void shift_table_deinit(shift_table_data_t table_data);
shift_table_data_t shift_table_init(int table_size); 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 typedef struct shift_addfast_data_s
{ {
float dsin[4]; float dsin[4];
float dcos[4]; float dcos[4];
float phase_increment; float phase_increment;
} shift_addfast_data_t; } shift_addfast_data_t;
shift_addfast_data_t shift_addfast_init(float rate); shift_addfast_data_t shift_addfast_init(float rate);
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 typedef struct shift_unroll_data_s
{ {
float* dsin; float* dsin;
float* dcos; float* dcos;
float phase_increment; float phase_increment;
int size; int size;
} shift_unroll_data_t; } shift_unroll_data_t;
float shift_unroll_cc(complexf *input, complexf* output, int input_size, shift_unroll_data_t* d, float starting_phase); 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); 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 typedef struct rtty_baudot_item_s
{ {
unsigned long long code; unsigned long long code;
unsigned char ascii_letter; unsigned char ascii_letter;
unsigned char ascii_figure; unsigned char ascii_figure;
} rtty_baudot_item_t; } rtty_baudot_item_t;
typedef enum rtty_baudot_decoder_state_e typedef enum rtty_baudot_decoder_state_e
{ {
RTTY_BAUDOT_WAITING_STOP_PULSE = 0, RTTY_BAUDOT_WAITING_STOP_PULSE = 0,
RTTY_BAUDOT_WAITING_START_PULSE, RTTY_BAUDOT_WAITING_START_PULSE,
RTTY_BAUDOT_RECEIVING_DATA RTTY_BAUDOT_RECEIVING_DATA
} rtty_baudot_decoder_state_t; } rtty_baudot_decoder_state_t;
typedef struct rtty_baudot_decoder_s typedef struct rtty_baudot_decoder_s
{ {
unsigned char fig_mode; unsigned char fig_mode;
unsigned char character_received; unsigned char character_received;
unsigned short shr; unsigned short shr;
unsigned char bit_cntr; unsigned char bit_cntr;
rtty_baudot_decoder_state_t state; rtty_baudot_decoder_state_t state;
} rtty_baudot_decoder_t; } rtty_baudot_decoder_t;
#define RTTY_FIGURE_MODE_SELECT_CODE 0b11011 #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 typedef struct psk31_varicode_item_s
{ {
unsigned long long code; unsigned long long code;
int bitcount; int bitcount;
unsigned char ascii; unsigned char ascii;
} psk31_varicode_item_t; } psk31_varicode_item_t;
char psk31_varicode_decoder_push(unsigned long long* status_shr, unsigned char symbol); 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 typedef struct serial_line_s
{ {
float samples_per_bits; float samples_per_bits;
int databits; //including parity int databits; //including parity
float stopbits; float stopbits;
int output_size; int output_size;
int input_used; int input_used;
float bit_sampling_width_ratio; float bit_sampling_width_ratio;
} serial_line_t; } serial_line_t;
void serial_line_decoder_f_u8(serial_line_t* s, float* input, unsigned char* output, int input_size); 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 typedef enum pll_type_e
{ {
PLL_P_CONTROLLER=1, PLL_P_CONTROLLER=1,
PLL_PI_CONTROLLER=2 PLL_PI_CONTROLLER=2
} pll_type_t; } pll_type_t;
typedef struct pll_s typedef struct pll_s
{ {
pll_type_t pll_type; pll_type_t pll_type;
//common: //common:
float output_phase; float output_phase;
float dphase; float dphase;
float frequency; float frequency;
float alpha; float alpha;
float beta; float beta;
float iir_temp; float iir_temp;
} pll_t; } pll_t;
void pll_cc_init_pi_controller(pll_t* p, float bandwidth, float ko, float kd, float damping_factor); 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 typedef enum timing_recovery_algorithm_e
{ {
TIMING_RECOVERY_ALGORITHM_GARDNER, TIMING_RECOVERY_ALGORITHM_GARDNER,
TIMING_RECOVERY_ALGORITHM_EARLYLATE TIMING_RECOVERY_ALGORITHM_EARLYLATE
} timing_recovery_algorithm_t; } timing_recovery_algorithm_t;
#define TIMING_RECOVERY_ALGORITHM_DEFAULT TIMING_RECOVERY_ALGORITHM_GARDNER #define TIMING_RECOVERY_ALGORITHM_DEFAULT TIMING_RECOVERY_ALGORITHM_GARDNER
typedef struct timing_recovery_state_s typedef struct timing_recovery_state_s
{ {
timing_recovery_algorithm_t algorithm; timing_recovery_algorithm_t algorithm;
int decimation_rate; // = input_rate / output_rate. We should get an input signal that is N times oversampled. int decimation_rate; // = input_rate / output_rate. We should get an input signal that is N times oversampled.
int output_size; int output_size;
int input_processed; int input_processed;
int use_q; //use both I and Q for calculating the error int use_q; //use both I and Q for calculating the error
int debug_phase; int debug_phase;
int debug_count; int debug_count;
int debug_force; int debug_force;
int debug_writefiles; int debug_writefiles;
int last_correction_offset; int last_correction_offset;
float earlylate_ratio; float earlylate_ratio;
} timing_recovery_state_t; } timing_recovery_state_t;
timing_recovery_state_t timing_recovery_init(timing_recovery_algorithm_t algorithm, int decimation_rate, int use_q); 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 typedef struct bpsk_costas_loop_state_s
{ {
float rc_filter_alpha; float rc_filter_alpha;
float vco_phase_addition_multiplier; float vco_phase_addition_multiplier;
float vco_phase; float vco_phase;
float last_lpfi_output; float last_lpfi_output;
float last_lpfq_output; float last_lpfq_output;
float last_vco_phase_addition; float last_vco_phase_addition;
} bpsk_costas_loop_state_t; } bpsk_costas_loop_state_t;
bpsk_costas_loop_state_t init_bpsk_costas_loop_cc(float samples_per_bits); bpsk_costas_loop_state_t init_bpsk_costas_loop_cc(float samples_per_bits);