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Added a squelch

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ha7ilm 2016-03-20 16:41:37 +01:00
parent 43e554da37
commit 4230198d91
4 changed files with 335 additions and 256 deletions
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54
README.md
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@ -36,7 +36,7 @@ Usage by example
- 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`.
- We want to listen one radio station at the frequency `-f 89500000` Hz (89.5 MHz).
- No other radio station is within the sampled bandwidth, so we send the signal directly to the demodulator. (This is an easy, but not perfect solution as the anti-aliasing filter at RTL-SDR DDC is too short.)
- No other radio station is within the sampled bandwidth, so we send the signal directly to the demodulator. (This is an easy, but not perfect solution as the anti-aliasing filter at RTL-SDR DDC is too short.)
- After FM demodulation we decimate the signal by a factor of 5 to match the rate of the audio card (240000 / 5 = 48000).
- A de-emphasis filter is used, because pre-emphasis is applied at the transmitter to compensate noise at higher frequencies. The time constant for de-emphasis for FM broadcasting in Europe is 50 microseconds (hence the `50e-6`).
- Also, `mplayer` cannot play floating point audio, so we convert our signal to a stream of 16-bit integers.
@ -58,12 +58,12 @@ Sample rates look like this:
*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.
### 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_i16 | 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).
@ -74,7 +74,7 @@ The first parameter is the frequency in MHz, and the second optional parameter i
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_i16 | 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.
### Design FIR band-pass filter (with complex taps)
@ -89,14 +89,14 @@ The first parameter is the frequency in MHz, and the second optional parameter i
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_i16 | 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
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.)
- 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.
Usage
-----
@ -107,7 +107,7 @@ Function name endings found in *libcsdr* mean the input and output data types of
Data types are noted as it follows:
- `f` is `float` (single percision)
- `c` is `complexf` (two single precision floating point values in a struct)
- `c` is `complexf` (two single precision floating point values in a struct)
- `u8` is `unsigned char` of 1 byte/8 bits (e. g. the output of `rtl_sdr` is of `u8`)
- `s16` is `signed short` of 2 bytes/16 bits (e. g. sound card input is usually `s16`)
@ -117,14 +117,14 @@ Functions usually end as:
- `_cf` complex input, float output
- `_cc` complex input, complex output
Regarding *csdr*, it can convert a real/complex stream from one data format to another, to interface it with other SDR tools and the sound card.
The following commands are available:
Regarding *csdr*, it can convert a real/complex stream from one data format to another, to interface it with other SDR tools and the sound card.
The following commands are available:
- `csdr convert_u8_f`
- `csdr convert_f_u8`
- `csdr convert_u8_f`
- `csdr convert_f_u8`
- `csdr convert_s8_f`
- `csdr convert_f_s8`
- `csdr convert_s16_f`
- `csdr convert_s16_f`
- `csdr convert_f_s16`
How to interpret: `csdr convert_<src>_<dst>`
@ -169,7 +169,7 @@ The `csdr` process just exits with 0.
yes_f <to_repeat> [buf_times]
It outputs continously the `to_repeat` float number.
It outputs continously the `to_repeat` float number.
If `buf_times` is not given, it never stops.
Else, after outputing `buf_times` number of buffers (the size of which is stated in the `BUFSIZE` macro), it exits.
@ -179,7 +179,7 @@ Along with copying its input samples to the output, it prints a warning message
floatdump_f
It prints any floating point input samples.
It prints any floating point input samples.
The format string used is `"%g "`.
flowcontrol <data_rate> <reads_per_second>
@ -190,8 +190,8 @@ It copies `data_rate / reads_per_second` bytes from the input to the output, doi
shift_math_cc <rate>
It shifts the signal in the frequency domain by `rate`.
`rate` is a floating point number between -0.5 and 0.5.
`rate` is relative to the sampling rate.
`rate` is a floating point number between -0.5 and 0.5.
`rate` is relative to the sampling rate.
Internally, a sine and cosine wave is generated to perform this function, and this function uses `math.h` for this purpose, which is quite accurate, but not always very fast.
@ -249,7 +249,7 @@ It uses fixed filters so it works only on predefined sample rates, for the actua
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.
amdemod_estimator_cf
@ -278,7 +278,7 @@ Other parameters were explained above at `firdes_lowpass_f`.
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.
To avoid aliasing, it runs a filter on the signal and removes spectral components above `0.5 × nyquist_frequency × decimation_factor`.
`transition_bw` and `window` are the parameters of the filter.
@ -304,7 +304,7 @@ Parameters are described under `firdes_bandpass_c` and `firdes_lowpass_f`.
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.
- `hang_time` is the number of samples to wait before strating to increase the gain after a peak.
- `reference` is the reference level for the AGC. It tries to keep the amplitude of the output signal close to that.
@ -322,7 +322,7 @@ It is a faster AGC that linearly changes the gain, taking the highest amplitude
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.
It can draw the spectrum by using `--octave`, for more information, look at the [Usage by example] section.
@ -357,7 +357,7 @@ It exchanges the first and second part of the FFT vector, to prepare it for the
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.
It does so by generating a complex signal:
* the real part of which is the input real signal,
* the imaginary part of which is `q_value` (0 by default).
@ -387,6 +387,10 @@ It doubles every input sample.
See the [buffer sizes](#buffer_sizes) section.
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.
#### Control via pipes
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.
@ -406,7 +410,7 @@ Processing will only start after the first control command has been received by
By writing to the given FIFO file with the syntax below, you can control the shift rate:
<low_cut> <high_cut>\n
E.g. you can send `-0.05 0.02\n`
#### Buffer sizes
@ -416,12 +420,12 @@ E.g. you can send `-0.05 0.02\n`
* *dynamic buffer size determination:* input buffer size is recommended by the previous process, output buffer size is determined by the process,
* *fixed buffer sizes*.
*csdr* can choose from two different buffer sizes by **default**.
*csdr* can choose from two different buffer sizes by **default**.
* For operations handling the full-bandwidth I/Q data from the receiver, a buffer size of 16384 samples is used (see `env_csdr_fixed_big_bufsize` in the code).
* For operations handling only a selected channel, a buffer size of 1024 samples is used (see `env_csdr_fixed_bufsize` in the code).
*csdr* now has an experimental feature called **dynamic buffer size determination**, which is switched on by issuing `export CSDR_DYNAMIC_BUFSIZE_ON=1` in the shell before running `csdr`. If it is enabled:
* All `csdr` processes in a DSP chain acquire their recommended input buffer size from the previous `csdr` process. This information is in the first 8 bytes of the input stream.
* All `csdr` processes in a DSP chain acquire their recommended input buffer size from the previous `csdr` process. This information is in the first 8 bytes of the input stream.
* Each process can decide whether to use this or choose another input buffer size (if that's more practical).
* Every process sends out its output buffer size to the next process. Then it startss processing data.
* The DSP chain should start with a `csdr setbuf <buffer_size>` process, which only copies data from the input to the output, but also sends out the given buffer size information to the next process.
@ -451,7 +455,7 @@ Example of initalization if the process generates N/D output samples for N input
Example of initialization if the process allocates memory for itself, and it doesn't want to use the global buffers:
getbufsize(); //dummy
getbufsize(); //dummy
sendbufsize(my_own_bufsize);
Example of initialization if the process always works with a fixed output size, regardless of the input:

301
csdr.c
View file

@ -1,5 +1,5 @@
/*
This software is part of libcsdr, a set of simple DSP routines for
This software is part of libcsdr, a set of simple DSP routines for
Software Defined Radio.
Copyright (c) 2014, Andras Retzler <randras@sdr.hu>
@ -35,14 +35,14 @@ SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#include <stdlib.h>
#include <string.h>
#include <limits.h>
#include <sys/time.h>
#include <sys/time.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <unistd.h>
#include <time.h>
#include <stdarg.h>
#include <stdarg.h>
#include "libcsdr.h"
#include "libcsdr_gpl.h"
#include "ima_adpcm.h"
@ -71,7 +71,9 @@ char usage[]=
" floatdump_f\n"
" flowcontrol <data_rate> <reads_per_second> [prebuffer_sec] [thrust]\n"
" shift_math_cc <rate>\n"
" shift_math_cc --fifo <fifo_path>\n"
" shift_addition_cc <rate>\n"
" shift_addition_cc --fifo <fifo_path>\n"
" shift_addition_cc_test\n"
" shift_table_cc <rate> [table_size]\n"
" decimating_shift_addition_cc <rate> [decimation]\n"
@ -95,6 +97,7 @@ char usage[]=
" logpower_cf [add_db]\n"
" fft_benchmark <fft_size> <fft_cycles> [--benchmark]\n"
" bandpass_fir_fft_cc <low_cut> <high_cut> <transition_bw> [window]\n"
" bandpass_fir_fft_cc --fifo <fifo_path> <transition_bw> [window]\n"
" encode_ima_adpcm_s16_u8\n"
" decode_ima_adpcm_u8_s16\n"
" compress_fft_adpcm_f_u8 <fft_size>\n"
@ -107,6 +110,7 @@ char usage[]=
" monos2stereo_s16\n"
" setbuf <buffer_size>\n"
" fft_exchange_sides_ff <fft_size>\n"
" squelch_and_smeter_cc --fifo <squelch_fifo> --outfifo <smeter_fifo> <use_every_nth> <report_every_nth>\n"
" \n"
;
@ -192,12 +196,12 @@ int read_fifo_ctl(int fd, char* format, ...)
static int buffer_index=0;
int bytes_read=read(fd,buffer+buffer_index,(RFCTL_BUFSIZE-buffer_index)*sizeof(char));
if(bytes_read<=0) return 0;
int prev_newline_at=0;
int last_newline_at=0;
for(int i=0;i<buffer_index+bytes_read;i++)
for(int i=0;i<buffer_index+bytes_read;i++)
{
if(buffer[i]=='\n')
if(buffer[i]=='\n')
{
prev_newline_at=last_newline_at;
last_newline_at=i+1;
@ -238,8 +242,8 @@ int getbufsize()
}
float* input_buffer;
unsigned char* buffer_u8;
float* input_buffer;
unsigned char* buffer_u8;
float *output_buffer;
short *buffer_i16;
float *temp_f;
@ -284,7 +288,7 @@ int sendbufsize(int size)
int parse_env()
{
char* envtmp;
envtmp=getenv("CSDR_DYNAMIC_BUFSIZE_ON");
envtmp=getenv("CSDR_DYNAMIC_BUFSIZE_ON");
//fprintf(stderr, "envtmp: %s\n",envtmp);
if(envtmp)
{
@ -293,21 +297,21 @@ int parse_env()
}
else
{
envtmp=getenv("CSDR_FIXED_BUFSIZE");
if(envtmp)
envtmp=getenv("CSDR_FIXED_BUFSIZE");
if(envtmp)
{
env_csdr_fixed_big_bufsize = env_csdr_fixed_bufsize = atoi(envtmp);
}
}
envtmp=getenv("CSDR_PRINT_BUFSIZES");
if(envtmp)
if(envtmp)
{
env_csdr_print_bufsizes = atoi(envtmp);
}
}
int main(int argc, char *argv[])
{
{
parse_env();
argv_global=argv;
if(argc<=1) return badsyntax(0);
@ -319,9 +323,9 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"setbuf"))
{
if(argc<=2) return badsyntax("need required parameter (buffer size)");
if(argc<=2) return badsyntax("need required parameter (buffer size)");
sscanf(argv[2],"%d",&the_bufsize);
if(the_bufsize<=0) return badsyntax("buffer size <= 0 is invalid");
if(the_bufsize<=0) return badsyntax("buffer size <= 0 is invalid");
sendbufsize(the_bufsize);
clone_(the_bufsize); //After sending the buffer size out, just copy stdin to stdout
}
@ -430,9 +434,9 @@ int main(int argc, char *argv[])
}
if(!strcmp(argv[1],"gain_ff"))
{
if(argc<=2) return badsyntax("need required parameter (gain)");
if(argc<=2) return badsyntax("need required parameter (gain)");
float gain;
sscanf(argv[2],"%g",&gain);
sscanf(argv[2],"%g",&gain);
if(!sendbufsize(initialize_buffers())) return -2;
for(;;)
{
@ -459,23 +463,23 @@ int main(int argc, char *argv[])
}
if(!strcmp(argv[1],"yes_f"))
{
if(argc<=2) return badsyntax("need required parameter (to_repeat)");
if(argc<=2) return badsyntax("need required parameter (to_repeat)");
float to_repeat;
sscanf(argv[2],"%g",&to_repeat);
int buf_times = 0;
if(argc>=4) sscanf(argv[3],"%d",&buf_times);
if(!sendbufsize(initialize_buffers())) return -2;
for(int i=0;i<the_bufsize;i++) output_buffer[i]=to_repeat;
for(int i=0;(!buf_times)||i<buf_times;i++)
{
fwrite(output_buffer, sizeof(float), the_bufsize, stdout);
TRY_YIELD;
for(int i=0;(!buf_times)||i<buf_times;i++)
{
fwrite(output_buffer, sizeof(float), the_bufsize, stdout);
TRY_YIELD;
}
return 0;
}
if(!strcmp(argv[1],"shift_math_cc"))
{
if(argc<=2) return badsyntax("need required parameter (rate)");
if(argc<=2) return badsyntax("need required parameter (rate)");
float starting_phase=0;
float rate;
sscanf(argv[2],"%g",&rate);
@ -489,7 +493,7 @@ int main(int argc, char *argv[])
}
return 0;
}
//speed tests:
//speed tests:
//csdr yes_f 1 1000000 | time csdr shift_math_cc 0.2 >/dev/null
//csdr yes_f 1 1000000 | time csdr shift_addition_cc 0.2 >/dev/null
//csdr yes_f 1 1000000 | time csdr shift_table_cc 0.2 >/dev/null
@ -497,7 +501,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"shift_table_cc"))
{
bigbufs=1;
if(argc<=2) return badsyntax("need required parameter (rate)");
if(argc<=2) return badsyntax("need required parameter (rate)");
float starting_phase=0;
float rate;
int table_size=65536;
@ -521,7 +525,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"decimating_shift_addition_cc"))
{
bigbufs=1;
if(argc<=2) return badsyntax("need required parameter (rate)");
if(argc<=2) return badsyntax("need required parameter (rate)");
float starting_phase=0;
float rate;
int decimation=1;
@ -558,7 +562,7 @@ int main(int argc, char *argv[])
}
else
{
if(argc<=2) return badsyntax("need required parameter (rate)");
if(argc<=2) return badsyntax("need required parameter (rate)");
sscanf(argv[2],"%g",&rate);
}
@ -595,7 +599,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"shift_addition_cc_test"))
{
if(argc<=2) return badsyntax("need required parameter (rate)");
if(argc<=2) return badsyntax("need required parameter (rate)");
float rate;
sscanf(argv[2],"%g",&rate);
//if(initialize_buffers()) return -2; //most likely we don't need this here
@ -607,7 +611,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"dcblock_ff"))
{
static dcblock_preserve_t dcp; //will be 0 as .bss is set to 0
if(!sendbufsize(initialize_buffers())) return -2;
if(!sendbufsize(initialize_buffers())) return -2;
for(;;)
{
FEOF_CHECK;
@ -682,7 +686,7 @@ int main(int argc, char *argv[])
}
if(!strcmp(argv[1],"deemphasis_wfm_ff"))
{
if(argc<=3) return badsyntax("need required parameters (sample rate, tau)");
if(argc<=3) return badsyntax("need required parameters (sample rate, tau)");
if(!sendbufsize(initialize_buffers())) return -2;
int sample_rate;
sscanf(argv[2],"%d",&sample_rate);
@ -707,19 +711,19 @@ int main(int argc, char *argv[])
{
FEOF_CHECK;
FREAD_R;
int nan_detect=0;
for(int i=0; i<the_bufsize;i++)
int nan_detect=0;
for(int i=0; i<the_bufsize;i++)
{
if(is_nan(input_buffer[i]))
{
nan_detect=1;
break;
if(is_nan(input_buffer[i]))
{
nan_detect=1;
break;
}
}
if(nan_detect) fprintf(stderr, "detect_nan_f: NaN detected!\n");
fwrite(input_buffer, sizeof(float), the_bufsize, stdout);
TRY_YIELD;
}
}
}
if(!strcmp(argv[1],"floatdump_f"))
@ -732,23 +736,23 @@ int main(int argc, char *argv[])
for(int i=0; i<the_bufsize;i++) fprintf(stderr, "%g ",input_buffer[i]);
TRY_YIELD;
}
}
if(!strcmp(argv[1],"deemphasis_nfm_ff"))
{
if(argc<=2) return badsyntax("need required parameter (sample rate)");
if(argc<=2) return badsyntax("need required parameter (sample rate)");
int sample_rate;
sscanf(argv[2],"%d",&sample_rate);
if(!sendbufsize(initialize_buffers())) return -2; //maybe we should take a /2 of bufsize over here
int processed=0;
for(;;)
{
FEOF_CHECK;
fread(input_buffer+the_bufsize-processed, sizeof(float), processed, stdin);
processed=deemphasis_nfm_ff(input_buffer, output_buffer, the_bufsize, sample_rate);
if(!processed) return badsyntax("deemphasis_nfm_ff: invalid sample rate (this function works only with specific sample rates).");
if(!processed) return badsyntax("deemphasis_nfm_ff: invalid sample rate (this function works only with specific sample rates).");
memmove(input_buffer,input_buffer+processed,(the_bufsize-processed)*sizeof(float)); //memmove lets the source and destination overlap
fwrite(output_buffer, sizeof(float), processed, stdout);
TRY_YIELD;
@ -769,7 +773,7 @@ int main(int argc, char *argv[])
}
if(!strcmp(argv[1],"amdemod_estimator_cf"))
{
if(!sendbufsize(initialize_buffers())) return -2;
if(!sendbufsize(initialize_buffers())) return -2;
for(;;)
{
FEOF_CHECK;
@ -783,7 +787,7 @@ int main(int argc, char *argv[])
{
bigbufs=1;
if(argc<=2) return badsyntax("need required parameter (decimation factor)");
if(argc<=2) return badsyntax("need required parameter (decimation factor)");
int factor;
sscanf(argv[2],"%d",&factor);
@ -811,7 +815,7 @@ int main(int argc, char *argv[])
fprintf(stderr,"taps_length = %d\n", taps_length);
padded_taps_length = taps_length+(NEON_ALIGNMENT/4)-1 - ((taps_length+(NEON_ALIGNMENT/4)-1)%(NEON_ALIGNMENT/4));
fprintf(stderr,"padded_taps_length = %d\n", padded_taps_length);
taps = (float*) (float*)malloc(padded_taps_length+NEON_ALIGNMENT);
fprintf(stderr,"taps = %x\n", taps);
taps = (float*)((((unsigned)taps)+NEON_ALIGNMENT-1) & ~(NEON_ALIGNMENT-1));
@ -830,13 +834,13 @@ int main(int argc, char *argv[])
{
FEOF_CHECK;
output_size=fir_decimate_cc((complexf*)input_buffer, (complexf*)output_buffer, the_bufsize, factor, taps, padded_taps_length);
//fprintf(stderr, "os %d\n",output_size);
//fprintf(stderr, "os %d\n",output_size);
fwrite(output_buffer, sizeof(complexf), output_size, stdout);
fflush(stdout);
TRY_YIELD;
input_skip=factor*output_size;
memmove((complexf*)input_buffer,((complexf*)input_buffer)+input_skip,(the_bufsize-input_skip)*sizeof(complexf)); //memmove lets the source and destination overlap
fread(((complexf*)input_buffer)+(the_bufsize-input_skip), sizeof(complexf), input_skip, stdin);
fread(((complexf*)input_buffer)+(the_bufsize-input_skip), sizeof(complexf), input_skip, stdin);
//fprintf(stderr,"iskip=%d output_size=%d start=%x target=%x skipcount=%x \n",input_skip,output_size,input_buffer, ((complexf*)input_buffer)+(BIG_BUFSIZE-input_skip),(BIG_BUFSIZE-input_skip));
}
}
@ -857,7 +861,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"firdes_lowpass_f"))
{
//Process the params
if(argc<=3) return badsyntax("need required parameters (cutoff_rate, length)");
if(argc<=3) return badsyntax("need required parameters (cutoff_rate, length)");
float cutoff_rate;
sscanf(argv[2],"%g",&cutoff_rate);
@ -884,9 +888,9 @@ int main(int argc, char *argv[])
if(octave) printf("taps=[");
for(int i=0;i<length;i++) printf("%g ",taps[i]);
if(octave) printf("];plot(taps);figure(2);freqz(taps);\n");
//Wait forever, so that octave won't close just after popping up the window.
//Wait forever, so that octave won't close just after popping up the window.
//You can close it with ^C.
if(octave) { fflush(stdout); getchar(); }
return 0;
@ -894,7 +898,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"firdes_bandpass_c"))
{
//Process the params
if(argc<=4) return badsyntax("need required parameters (low_cut, high_cut, length)");
if(argc<=4) return badsyntax("need required parameters (low_cut, high_cut, length)");
float low_cut;
sscanf(argv[2],"%g",&low_cut);
@ -932,8 +936,8 @@ int main(int argc, char *argv[])
"subplot(2,1,2);plot(arg(fser));\n"
"#figure(2);freqz(taps);\n"
"#figur(3);plot3(taps);\n",fft_length-length);
//Wait forever, so that octave won't close just after popping up the window.
//Wait forever, so that octave won't close just after popping up the window.
//You can close it with ^C.
if(octave) { fflush(stdout); getchar(); }
return 0;
@ -983,12 +987,12 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"fastagc_ff"))
{
static fastagc_ff_t input; //is in .bss and gets cleared to zero before main()
static fastagc_ff_t input; //is in .bss and gets cleared to zero before main()
input.input_size=1024;
if(argc>=3) sscanf(argv[2],"%d",&input.input_size);
getbufsize(); //dummy
getbufsize(); //dummy
sendbufsize(input.input_size);
input.reference=1.0;
@ -996,7 +1000,7 @@ int main(int argc, char *argv[])
//input.max_peak_ratio=12.0;
//if(argc>=5) sscanf(argv[3],"%g",&input.max_peak_ratio);
input.buffer_1=(float*)calloc(input.input_size,sizeof(float));
input.buffer_2=(float*)calloc(input.input_size,sizeof(float));
input.buffer_input=(float*)malloc(sizeof(float)*input.input_size);
@ -1005,7 +1009,7 @@ int main(int argc, char *argv[])
{
FEOF_CHECK;
fread(input.buffer_input, sizeof(float), input.input_size, stdin);
fastagc_ff(&input, agc_output_buffer);
fastagc_ff(&input, agc_output_buffer);
fwrite(agc_output_buffer, sizeof(float), input.input_size, stdout);
TRY_YIELD;
}
@ -1014,11 +1018,11 @@ int main(int argc, char *argv[])
int suboptimal;
if( (suboptimal=!strcmp(argv[1],"suboptimal_rational_resampler_ff"))||(!strcmp(argv[1],"rational_resampler_ff")) )
{
//last@2014-11-06: ./docompile; ./csdr yes_f 1.0 | ./csdr suboptimal_rational_resampler_ff 5 2
//Process the params
if(argc<=3) return badsyntax("need required parameters (interpolation, decimation)");
if(argc<=3) return badsyntax("need required parameters (interpolation, decimation)");
int interpolation;
sscanf(argv[2],"%d",&interpolation);
int decimation;
@ -1050,7 +1054,7 @@ int main(int argc, char *argv[])
int taps_length = firdes_filter_len(transition_bw);
float* taps = (float*)malloc(sizeof(float)*taps_length);
rational_resampler_get_lowpass_f(taps, taps_length, interpolation, decimation, window);
static rational_resampler_ff_t d; //in .bss => initialized to zero
for(;;)
@ -1071,7 +1075,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"fractional_decimator_ff"))
{
//Process the params
if(argc<=2) return badsyntax("need required parameters (rate)");
if(argc<=2) return badsyntax("need required parameters (rate)");
float rate;
sscanf(argv[2],"%g",&rate);
@ -1086,7 +1090,7 @@ int main(int argc, char *argv[])
else fprintf(stderr,"fractional_decimator_ff: window = %s\n",firdes_get_string_from_window(window));
if(!initialize_buffers()) return -2;
sendbufsize(the_bufsize / rate);
sendbufsize(the_bufsize / rate);
if(rate==1) clone_(the_bufsize); //copy input to output in this special case (and stick in this function).
@ -1094,7 +1098,7 @@ int main(int argc, char *argv[])
int taps_length = firdes_filter_len(transition_bw);
fprintf(stderr,"fractional_decimator_ff: taps_length = %d\n",taps_length);
float* taps = (float*)malloc(sizeof(float)*taps_length);
firdes_lowpass_f(taps, taps_length, 0.59*0.5/(rate-transition_bw), window); //0.6 const to compensate rolloff
firdes_lowpass_f(taps, taps_length, 0.59*0.5/(rate-transition_bw), window); //0.6 const to compensate rolloff
//for(int=0;i<taps_length; i++) fprintf(stderr,"%g ",taps[i]);
static fractional_decimator_ff_t d; //in .bss => initialized to zero
@ -1112,10 +1116,10 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"fft_cc"))
{
if(argc<=3) return badsyntax("need required parameters (fft_size, out_of_every_n_samples)");
if(argc<=3) return badsyntax("need required parameters (fft_size, out_of_every_n_samples)");
int fft_size;
sscanf(argv[2],"%d",&fft_size);
if(log2n(fft_size)==-1) return badsyntax("fft_size should be power of 2");
if(log2n(fft_size)==-1) return badsyntax("fft_size should be power of 2");
int every_n_samples;
sscanf(argv[3],"%d",&every_n_samples);
int benchmark=0;
@ -1125,12 +1129,12 @@ int main(int argc, char *argv[])
{
window=firdes_get_window_from_string(argv[4]);
}
if(argc>=6)
if(argc>=6)
{
benchmark|=!strcmp("--benchmark",argv[5]);
octave|=!strcmp("--octave",argv[5]);
}
if(argc>=7)
if(argc>=7)
{
benchmark|=!strcmp("--benchmark",argv[6]);
octave|=!strcmp("--octave",argv[6]);
@ -1172,7 +1176,7 @@ int main(int argc, char *argv[])
printf("fftdata=[");
//we have to swap the two parts of the array to get a valid spectrum
for(int i=fft_size/2;i<fft_size;i++) printf("(%g)+(%g)*i ",iof(output,i),qof(output,i));
for(int i=0;i<fft_size/2;i++) printf("(%g)+(%g)*i ",iof(output,i),qof(output,i));
for(int i=0;i<fft_size/2;i++) printf("(%g)+(%g)*i ",iof(output,i),qof(output,i));
printf(
"];\n"
"y=abs(fftdata);\n"
@ -1189,7 +1193,7 @@ int main(int argc, char *argv[])
float add_db=0;
if(argc>=3) sscanf(argv[2],"%g",&add_db);
if(!sendbufsize(initialize_buffers())) return -2;
if(!sendbufsize(initialize_buffers())) return -2;
for(;;)
{
@ -1203,7 +1207,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"fft_exchange_sides_ff"))
{
if(argc<=2) return badsyntax("need required parameters (fft_size)");
if(argc<=2) return badsyntax("need required parameters (fft_size)");
int fft_size;
sscanf(argv[2],"%d",&fft_size);
if(!getbufsize()) return -2; //dummy
@ -1226,13 +1230,13 @@ int main(int argc, char *argv[])
#define COMPRESS_FFT_PAD_N 10
//We will pad the FFT at the beginning, with the first value of the input data, COMPRESS_FFT_PAD_N times.
//No, this is not advanced DSP, just the ADPCM codec produces some gabarge samples at the beginning,
//so we just add data to become garbage and get skipped.
//No, this is not advanced DSP, just the ADPCM codec produces some gabarge samples at the beginning,
//so we just add data to become garbage and get skipped.
//COMPRESS_FFT_PAD_N should be even.
if(!strcmp(argv[1],"compress_fft_adpcm_f_u8"))
{
if(argc<=2) return badsyntax("need required parameters (fft_size)");
if(argc<=2) return badsyntax("need required parameters (fft_size)");
int fft_size;
sscanf(argv[2],"%d",&fft_size);
int real_data_size=fft_size+COMPRESS_FFT_PAD_N;
@ -1260,12 +1264,12 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"fft_benchmark"))
{
if(argc<=3) return badsyntax("need required parameters (fft_size, fft_cycles)");
if(argc<=3) return badsyntax("need required parameters (fft_size, fft_cycles)");
int fft_size;
sscanf(argv[2],"%d",&fft_size);
int fft_cycles;
sscanf(argv[3],"%d",&fft_cycles);
int benchmark=(argc>=5)&&!strcmp(argv[4],"--benchmark");
fprintf(stderr,"fft_benchmark: FFT library used: %s\n",FFT_LIBRARY_USED);
@ -1274,20 +1278,20 @@ int main(int argc, char *argv[])
//fill input with random data
srand(time(NULL));
for(int i=0;i<fft_size;i++)
{
for(int i=0;i<fft_size;i++)
{
iof(input,i)=rand()/(float)INT_MAX;
qof(input,i)=rand()/(float)INT_MAX;
}
//initialize FFT library, and measure time
fprintf(stderr,"fft_benchmark: initializing... ");
struct timespec start_time, end_time;
struct timespec start_time, end_time;
clock_gettime(CLOCK_MONOTONIC_RAW, &start_time);
FFT_PLAN_T* plan=make_fft_c2c(fft_size,input,output,1,benchmark);
clock_gettime(CLOCK_MONOTONIC_RAW, &end_time);
fprintf(stderr,"done in %g seconds.\n",TIME_TAKEN(start_time,end_time));
//do the actual measurement about the FFT
clock_gettime(CLOCK_MONOTONIC_RAW, &start_time);
for(int i=0;i<fft_cycles;i++) fft_execute(plan);
@ -1296,15 +1300,15 @@ int main(int argc, char *argv[])
fprintf(stderr,"fft_benchmark: %d transforms of %d processed in %g seconds, %g seconds each.\n",fft_cycles,fft_size,time_taken_fft,time_taken_fft/fft_cycles);
return 0;
}
if(!strcmp(argv[1],"bandpass_fir_fft_cc")) //this command does not exist as a separate function
{
float low_cut;
float high_cut;
float transition_bw;
window_t window = WINDOW_DEFAULT;
char window_string[256]; //TODO: nice buffer overflow opportunity
char window_string[256]; //TODO: nice buffer overflow opportunity
int fd;
if(fd=init_fifo(argc,argv))
{
@ -1313,7 +1317,7 @@ int main(int argc, char *argv[])
}
else
{
if(argc<=4) return badsyntax("need required parameters (low_cut, high_cut, transition_bw)");
if(argc<=4) return badsyntax("need required parameters (low_cut, high_cut, transition_bw)");
sscanf(argv[2],"%g",&low_cut);
sscanf(argv[3],"%g",&high_cut);
}
@ -1325,14 +1329,14 @@ int main(int argc, char *argv[])
int taps_length=firdes_filter_len(transition_bw); //the number of non-zero taps
int fft_size=next_pow2(taps_length); //we will have to pad the taps with zeros until the next power of 2 for FFT
//the number of padding zeros is the number of output samples we will be able to take away after every processing step, and it looks sane to check if it is large enough.
if (fft_size-taps_length<200) fft_size<<=1;
if (fft_size-taps_length<200) fft_size<<=1;
int input_size = fft_size - taps_length + 1;
int overlap_length = taps_length - 1;
fprintf(stderr,"bandpass_fir_fft_cc: (fft_size = %d) = (taps_length = %d) + (input_size = %d) - 1\n(overlap_length = %d) = taps_length - 1\n", fft_size, taps_length, input_size, overlap_length);
if (fft_size<=2) return badsyntax("FFT size error.");
if(!sendbufsize(getbufsize())) return -2;
//prepare making the filter and doing FFT on it
complexf* taps=(complexf*)calloc(sizeof(complexf),fft_size); //initialize to zero
complexf* taps_fft=(complexf*)malloc(sizeof(complexf)*fft_size);
@ -1342,16 +1346,16 @@ int main(int argc, char *argv[])
complexf* input = fft_malloc(fft_size*sizeof(complexf));
complexf* input_fourier = fft_malloc(fft_size*sizeof(complexf));
FFT_PLAN_T* plan_forward = make_fft_c2c(fft_size, input, input_fourier, 1, 1); //forward, do benchmark
complexf* output_fourier = fft_malloc(fft_size*sizeof(complexf));
complexf* output_1 = fft_malloc(fft_size*sizeof(complexf));
complexf* output_2 = fft_malloc(fft_size*sizeof(complexf));
//we create 2x output buffers so that one will preserve the previous overlap:
FFT_PLAN_T* plan_inverse_1 = make_fft_c2c(fft_size, output_fourier, output_1, 0, 1); //inverse, do benchmark
FFT_PLAN_T* plan_inverse_2 = make_fft_c2c(fft_size, output_fourier, output_2, 0, 1);
//we initialize this buffer to 0 as it will be taken as the overlap source for the first time:
for(int i=0;i<fft_size;i++) iof(plan_inverse_2->output,i)=qof(plan_inverse_2->output,i)=0;
FFT_PLAN_T* plan_inverse_2 = make_fft_c2c(fft_size, output_fourier, output_2, 0, 1);
//we initialize this buffer to 0 as it will be taken as the overlap source for the first time:
for(int i=0;i<fft_size;i++) iof(plan_inverse_2->output,i)=qof(plan_inverse_2->output,i)=0;
for(int i=input_size;i<fft_size;i++) iof(input,i)=qof(input,i)=0; //we pre-pad the input buffer with zeros
for(;;)
@ -1410,10 +1414,10 @@ int main(int argc, char *argv[])
}
}
#endif
if(!strcmp(argv[1],"flowcontrol"))
{
if(argc<=3) return badsyntax("need required parameters (data_rate, reads_per_seconds)");
if(argc<=3) return badsyntax("need required parameters (data_rate, reads_per_seconds)");
int data_rate;
sscanf(argv[2],"%d",&data_rate);
int reads_per_second;
@ -1437,8 +1441,8 @@ int main(int argc, char *argv[])
#if 0
if(!strcmp(argv[1],"flowcontrol"))
{
if(argc<=3) return badsyntax("need required parameters (data_rate, reads_per_seconds)");
if(argc<=3) return badsyntax("need required parameters (data_rate, reads_per_seconds)");
int data_rate;
sscanf(argv[2],"%d",&data_rate);
@ -1479,7 +1483,7 @@ int main(int argc, char *argv[])
int read_return;
struct timespec start_time, end_time;
unsigned long long int all_bytes_written=0;
int test=0;
@ -1490,7 +1494,7 @@ int main(int argc, char *argv[])
fprintf(stderr, "r");
read_return=read(STDIN_FILENO, flowcontrol_buffer+flowcontrol_bufindex, sizeof(unsigned char) * (flowcontrol_bufsize-flowcontrol_bufindex) );
fprintf(stderr, "t");
if(read_return>0) flowcontrol_bufindex+=read_return;
if(read_return>0) flowcontrol_bufindex+=read_return;
if(flowcontrol_is_buffering)
@ -1537,7 +1541,7 @@ int main(int argc, char *argv[])
if(!strcmp(argv[1],"through"))
{
struct timespec start_time, end_time;
struct timespec start_time, end_time;
if(!sendbufsize(initialize_buffers())) return -2;
int time_now_sec=0;
@ -1545,21 +1549,21 @@ int main(int argc, char *argv[])
unsigned char* through_buffer;
through_buffer = (unsigned char*)malloc(the_bufsize*sizeof(float));
for(;;)
{
FEOF_CHECK;
fread(through_buffer, sizeof(float), the_bufsize, stdin);
if(!time_now_sec)
{
time_now_sec=1;
clock_gettime(CLOCK_MONOTONIC_RAW, &start_time);
}
else
if(!time_now_sec)
{
clock_gettime(CLOCK_MONOTONIC_RAW, &end_time);
time_now_sec=1;
clock_gettime(CLOCK_MONOTONIC_RAW, &start_time);
}
else
{
clock_gettime(CLOCK_MONOTONIC_RAW, &end_time);
float timetaken;
if(time_now_sec<(timetaken=TIME_TAKEN(start_time,end_time)))
{
@ -1583,20 +1587,20 @@ int main(int argc, char *argv[])
{
FEOF_CHECK;
FREAD_R;
for(int i=0;i<the_bufsize;i++)
for(int i=0;i<the_bufsize;i++)
{
iof(output_buffer,i)=input_buffer[i];
qof(output_buffer,i)=q_value;
}
FWRITE_C;
TRY_YIELD;
}
}
}
if(!strcmp(argv[1],"convert_f_samplerf"))
{
if(argc<=2) return badsyntax("need required parameter (wait_for_this_sample)");
if(argc<=2) return badsyntax("need required parameter (wait_for_this_sample)");
unsigned wait_for_this_sample;
sscanf(argv[2],"%u",&wait_for_this_sample);
@ -1606,16 +1610,16 @@ int main(int argc, char *argv[])
{
FEOF_CHECK;
FREAD_R;
for(int i=0;i<the_bufsize;i++)
for(int i=0;i<the_bufsize;i++)
{
*((double*)(&samplerf_buf[16*i])) = input_buffer[i];
*((unsigned*)(&samplerf_buf[16*i+8])) = wait_for_this_sample;
*((unsigned*)(&samplerf_buf[16*i+12])) = 0;
}
fwrite(samplerf_buf, 16, the_bufsize, stdout);
TRY_YIELD;
}
}
}
if(!strcmp(argv[1],"add_dcoffset_cc"))
@ -1628,7 +1632,7 @@ int main(int argc, char *argv[])
add_dcoffset_cc((complexf*)input_buffer, (complexf*)output_buffer, the_bufsize);
FWRITE_C;
TRY_YIELD;
}
}
}
if(!strcmp(argv[1],"fmmod_fc"))
@ -1642,13 +1646,13 @@ int main(int argc, char *argv[])
last_phase = fmmod_fc(input_buffer, (complexf*)output_buffer, the_bufsize, last_phase);
FWRITE_C;
TRY_YIELD;
}
}
}
if(!strcmp(argv[1],"fixed_amplitude_cc"))
{
if(argc<=2) return badsyntax("need required parameter (new_amplitude)");
if(argc<=2) return badsyntax("need required parameter (new_amplitude)");
float new_amplitude;
sscanf(argv[2],"%g",&new_amplitude);
@ -1660,7 +1664,7 @@ int main(int argc, char *argv[])
fixed_amplitude_cc((complexf*)input_buffer, (complexf*)output_buffer, the_bufsize, new_amplitude);
FWRITE_C;
TRY_YIELD;
}
}
}
if((!strcmp(argv[1],"mono2stereo_i16"))||(!strcmp(argv[1],"mono2stereo_s16")))
@ -1671,23 +1675,74 @@ int main(int argc, char *argv[])
{
FEOF_CHECK;
fread (input_buffer, sizeof(short), the_bufsize, stdin);
for(int i=0;i<the_bufsize;i++)
for(int i=0;i<the_bufsize;i++)
{
*(((short*)output_buffer)+2*i)=*(((short*)input_buffer)+i);
*(((short*)output_buffer)+2*i+1)=*(((short*)input_buffer)+i);
}
fwrite (output_buffer, sizeof(short)*2, the_bufsize, stdout);
TRY_YIELD;
}
}
}
if(!strcmp(argv[1],"squelch_and_smeter_cc"))
{
if(!sendbufsize(initialize_buffers())) return -2;
float power;
float squelch_level;
int decimation;
int report_every_nth;
int fd;
char power_value_buf[101];
int power_value_buf_size;
int report_cntr=0;
complexf* zerobuf = (complexf*)malloc(sizeof(complexf)*the_bufsize);
for(int i=0;i<the_bufsize*2;i++) *(((float*)zerobuf)+i)=0;
if(fd=init_fifo(argc,argv)) while(!read_fifo_ctl(fd,"%g\n",&squelch_level)) usleep(10000);
else return badsyntax("need required parameter (--fifo <fifo>)");
fprintf(stderr, "squelch_and_power_cc: initial squelch level is %g\n", squelch_level);
if((argc<=5)||((argc>5)&&(strcmp(argv[4],"--outfifo")))) return badsyntax("need required parameter (--outfifo <fifo>)");
int fd2 = open(argv[5], O_WRONLY);
if(fd2==-1) return badsyntax("error while opening --outfifo");
int flags = fcntl(fd2, F_GETFL, 0);
fcntl(fd2, F_SETFL, flags | O_NONBLOCK);
if(argc<=6) return badsyntax("need required parameter (use_every_nth)");
sscanf(argv[6],"%d",&decimation);
if(decimation<=0) return badsyntax("use_every_nth <= 0 is invalid");
if(argc<=7) return badsyntax("need required parameter (report_every_nth)");
sscanf(argv[7],"%d",&report_every_nth);
if(report_every_nth<=0) return badsyntax("report_every_nth <= 0 is invalid");
for(;;)
{
FEOF_CHECK;
FREAD_C; //read input data
power = get_power_c((complexf*)input_buffer, the_bufsize, decimation);
if(report_cntr++>report_every_nth)
{
report_cntr=0;
power_value_buf_size=snprintf(power_value_buf,100,"%g\n",power);
write(fd2,power_value_buf,power_value_buf_size*sizeof(char));
}
if(squelch_level==0||power>=squelch_level)
{
//fprintf(stderr,"P");
fwrite(input_buffer, sizeof(complexf), the_bufsize, stdout);
}
else
{
//fprintf(stderr,"S");
fwrite(zerobuf, sizeof(complexf), the_bufsize, stdout);
}
if(read_fifo_ctl(fd,"%g\n",&squelch_level)) fprintf(stderr, "squelch_and_power_cc: new squelch level is %g\n", squelch_level);
TRY_YIELD;
}
}
if(!strcmp(argv[1],"none"))
{
return 0;
}
return badsyntax("function name given in argument 1 does not exist. Possible causes:\n- You mistyped the commandline.\n- You need to update csdr to a newer version (if available).");
}

210
libcsdr.c
View file

@ -1,5 +1,5 @@
/*
This software is part of libcsdr, a set of simple DSP routines for
This software is part of libcsdr, a set of simple DSP routines for
Software Defined Radio.
Copyright (c) 2014, Andras Retzler <randras@sdr.hu>
@ -40,14 +40,14 @@ SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#include <assert.h>
/*
_ _ __ _ _
(_) | | / _| | | (_)
__ ___ _ __ __| | _____ __ | |_ _ _ _ __ ___| |_ _ ___ _ __ ___
_ _ __ _ _
(_) | | / _| | | (_)
__ ___ _ __ __| | _____ __ | |_ _ _ _ __ ___| |_ _ ___ _ __ ___
\ \ /\ / / | '_ \ / _` |/ _ \ \ /\ / / | _| | | | '_ \ / __| __| |/ _ \| '_ \/ __|
\ V V /| | | | | (_| | (_) \ V V / | | | |_| | | | | (__| |_| | (_) | | | \__ \
\_/\_/ |_|_| |_|\__,_|\___/ \_/\_/ |_| \__,_|_| |_|\___|\__|_|\___/|_| |_|___/
*/
#define MFIRDES_GWS(NAME) \
@ -95,7 +95,7 @@ float firdes_wkernel_boxcar(float rate)
}
float (*firdes_get_window_kernel(window_t window))(float)
{
{
if(window==WINDOW_HAMMING) return firdes_wkernel_hamming;
else if(window==WINDOW_BLACKMAN) return firdes_wkernel_blackman;
else if(window==WINDOW_BOXCAR) return firdes_wkernel_boxcar;
@ -103,14 +103,14 @@ float (*firdes_get_window_kernel(window_t window))(float)
}
/*
______ _____ _____ __ _ _ _ _ _
| ____|_ _| __ \ / _(_) | | | | (_)
| |__ | | | |__) | | |_ _| | |_ ___ _ __ __| | ___ ___ _ __ _ _ __
| __| | | | _ / | _| | | __/ _ \ '__| / _` |/ _ \/ __| |/ _` | '_ \
______ _____ _____ __ _ _ _ _ _
| ____|_ _| __ \ / _(_) | | | | (_)
| |__ | | | |__) | | |_ _| | |_ ___ _ __ __| | ___ ___ _ __ _ _ __
| __| | | | _ / | _| | | __/ _ \ '__| / _` |/ _ \/ __| |/ _` | '_ \
| | _| |_| | \ \ | | | | | || __/ | | (_| | __/\__ \ | (_| | | | |
|_| |_____|_| \_\ |_| |_|_|\__\___|_| \__,_|\___||___/_|\__, |_| |_|
__/ |
|___/
__/ |
|___/
*/
void firdes_lowpass_f(float *output, int length, float cutoff_rate, window_t window)
@ -127,7 +127,7 @@ void firdes_lowpass_f(float *output, int length, float cutoff_rate, window_t win
output[middle-i]=output[middle+i]=(sin(2*PI*cutoff_rate*i)/i)*window_function((float)i/middle);
//printf("%g %d %d %d %d | %g\n",output[middle-i],i,middle,middle+i,middle-i,sin(2*PI*cutoff_rate*i));
}
//Normalize filter kernel
float sum=0;
for(int i=0;i<length;i++) //@firdes_lowpass_f: normalize pass 1
@ -147,18 +147,18 @@ void firdes_bandpass_c(complexf *output, int length, float lowcut, float highcut
// 2. we shift the filter taps spectrally by multiplying with e^(j*w), so we get complex taps
//(tnx HA5FT)
float* realtaps = (float*)malloc(sizeof(float)*length);
firdes_lowpass_f(realtaps, length, (highcut-lowcut)/2, window);
float filter_center=(highcut+lowcut)/2;
float phase=0, sinval, cosval;
for(int i=0; i<length; i++) //@@firdes_bandpass_c
{
cosval=cos(phase);
sinval=sin(phase);
sinval=sin(phase);
phase+=2*PI*filter_center;
while(phase>2*PI) phase-=2*PI; //@@firdes_bandpass_c
while(phase<0) phase+=2*PI;
while(phase<0) phase+=2*PI;
iof(output,i)=cosval*realtaps[i];
qof(output,i)=sinval*realtaps[i];
//output[i] := realtaps[i] * e^j*w
@ -173,14 +173,14 @@ int firdes_filter_len(float transition_bw)
}
/*
_____ _____ _____ __ _ _
| __ \ / ____| __ \ / _| | | (_)
| | | | (___ | |__) | | |_ _ _ _ __ ___| |_ _ ___ _ __ ___
_____ _____ _____ __ _ _
| __ \ / ____| __ \ / _| | | (_)
| | | | (___ | |__) | | |_ _ _ _ __ ___| |_ _ ___ _ __ ___
| | | |\___ \| ___/ | _| | | | '_ \ / __| __| |/ _ \| '_ \/ __|
| |__| |____) | | | | | |_| | | | | (__| |_| | (_) | | | \__ \
|_____/|_____/|_| |_| \__,_|_| |_|\___|\__|_|\___/|_| |_|___/
*/
*/
float shift_math_cc(complexf *input, complexf* output, int input_size, float rate, float starting_phase)
{
@ -239,7 +239,7 @@ float shift_table_cc(complexf* input, complexf* output, int input_size, float ra
//float vphase=fmodf(phase,PI/2); //between 0 and 90deg
int quadrant=phase/(PI/2); //between 0 and 3
float vphase=phase-quadrant*(PI/2);
sin_index=(vphase/(PI/2))*table_data.table_size;
sin_index=(vphase/(PI/2))*table_data.table_size;
cos_index=table_data.table_size-1-sin_index;
if(quadrant&1) //in quadrant 1 and 3
{
@ -280,8 +280,8 @@ int fir_decimate_cc(complexf *input, complexf *output, int input_size, int decim
for(int i=0; i<input_size; i+=decimation) //@fir_decimate_cc: outer loop
{
if(i+taps_length>input_size) break;
register float acci=0;
register float accq=0;
register float acci=0;
register float accq=0;
register int ti=0;
register float* pinput=(float*)&(input[i+ti]);
@ -289,10 +289,10 @@ int fir_decimate_cc(complexf *input, complexf *output, int input_size, int decim
register float* ptaps_end=taps+taps_length;
float quad_acciq [8];
/*
q0, q1: input signal I sample and Q sample
q2: taps
q2: taps
q4, q5: accumulator for I branch and Q branch (will be the output)
*/
@ -300,7 +300,7 @@ q4, q5: accumulator for I branch and Q branch (will be the output)
" vmov.f32 q4, #0.0\n\t" //another way to null the accumulators
" vmov.f32 q5, #0.0\n\t"
"for_fdccasm: vld2.32 {q0-q1}, [%[pinput]]!\n\t" //load q0 and q1 directly from the memory address stored in pinput, with interleaving (so that we get the I samples in q0 and the Q samples in q1), also increment the memory address in pinput (hence the "!" mark) //http://community.arm.com/groups/processors/blog/2010/03/17/coding-for-neon--part-1-load-and-stores
" vld1.32 {q2}, [%[ptaps]]!\n\t"
" vld1.32 {q2}, [%[ptaps]]!\n\t"
" vmla.f32 q4, q0, q2\n\t" //quad_acc_i += quad_input_i * quad_taps_1 //http://stackoverflow.com/questions/3240440/how-to-use-the-multiply-and-accumulate-intrinsics-in-arm-cortex-a8 //http://infocenter.arm.com/help/index.jsp?topic=/com.arm.doc.dui0489e/CIHEJBIE.html
" vmla.f32 q5, q1, q2\n\t" //quad_acc_q += quad_input_q * quad_taps_1
" cmp %[ptaps], %[ptaps_end]\n\t" //if(ptaps == ptaps_end)
@ -311,13 +311,13 @@ q4, q5: accumulator for I branch and Q branch (will be the output)
[pinput]"+r"(pinput), [ptaps]"+r"(ptaps) //output operand list
:
[ptaps_end]"r"(ptaps_end), [quad_acci]"r"(quad_acciq), [quad_accq]"r"(quad_acciq+4) //input operand list
:
:
"memory", "q0", "q1", "q2", "q4", "q5", "cc" //clobber list
);
//original for loops for reference:
//for(int ti=0; ti<taps_length; ti++) acci += (iof(input,i+ti)) * taps[ti]; //@fir_decimate_cc: i loop
//for(int ti=0; ti<taps_length; ti++) acci += (iof(input,i+ti)) * taps[ti]; //@fir_decimate_cc: i loop
//for(int ti=0; ti<taps_length; ti++) accq += (qof(input,i+ti)) * taps[ti]; //@fir_decimate_cc: q loop
//for(int n=0;n<8;n++) fprintf(stderr, "\n>> [%d] %g \n", n, quad_acciq[n]);
iof(output,oi)=quad_acciq[0]+quad_acciq[1]+quad_acciq[2]+quad_acciq[3]; //we're still not ready, as we have to add up the contents of a quad accumulator register to get a single accumulated value
qof(output,oi)=quad_acciq[4]+quad_acciq[5]+quad_acciq[6]+quad_acciq[7];
@ -381,7 +381,7 @@ int fir_decimate_cc(complexf *input, complexf *output, int input_size, int decim
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)
{
//Theory: http://www.dspguru.com/dsp/faqs/multirate/resampling
//oi: output index, i: tap index
int output_size=input_size*interpolation/decimation;
@ -412,7 +412,7 @@ rational_resampler_ff_t rational_resampler_ff(float *input, float *output, int i
/*
The greatest challenge in resampling is figuring out which tap should be applied to which sample.
The greatest challenge in resampling is figuring out which tap should be applied to which sample.
Typical test patterns for rational_resampler_ff:
@ -457,13 +457,13 @@ float inline fir_one_pass_ff(float* input, float* taps, int taps_length)
fractional_decimator_ff_t fractional_decimator_ff(float* input, float* output, int input_size, float rate, float *taps, int taps_length, fractional_decimator_ff_t d)
{
if(rate<=1.0) return d; //sanity check, can't decimate <=1.0
//This routine can handle floating point decimation rates.
//It linearly interpolates between two samples that are taken into consideration from the filtered input.
//This routine can handle floating point decimation rates.
//It linearly interpolates between two samples that are taken into consideration from the filtered input.
int oi=0;
int index_high;
float where=d.remain;
float result_high, result_low;
if(where==0.0) //in the first iteration index_high may be zero (so using the item index_high-1 would lead to invalid memory access).
if(where==0.0) //in the first iteration index_high may be zero (so using the item index_high-1 would lead to invalid memory access).
{
output[oi++]=fir_one_pass_ff(input,taps,taps_length);
where+=rate;
@ -474,7 +474,7 @@ fractional_decimator_ff_t fractional_decimator_ff(float* input, float* output, i
for(;(index_high=ceilf(where))+taps_length<input_size;where+=rate) //@fractional_decimator_ff
{
if(previous_index_high==index_high-1) result_low=result_high; //if we step less than 2.0 then we do already have the result for the FIR filter for that index
else result_low=fir_one_pass_ff(input+index_high-1,taps,taps_length);
else result_low=fir_one_pass_ff(input+index_high-1,taps,taps_length);
result_high=fir_one_pass_ff(input+index_high,taps,taps_length);
float register rate_between_samples=where-index_high+1;
output[oi++]=result_low*(1-rate_between_samples)+result_high*rate_between_samples;
@ -498,16 +498,16 @@ void apply_fir_fft_cc(FFT_PLAN_T* plan, FFT_PLAN_T* plan_inverse, complexf* taps
//multiply the filter and the input
complexf* in = plan->output;
complexf* out = plan_inverse->input;
for(int i=0;i<plan->size;i++) //@apply_fir_fft_cc: multiplication
{
iof(out,i)=iof(in,i)*iof(taps_fft,i)-qof(in,i)*qof(taps_fft,i);
qof(out,i)=iof(in,i)*qof(taps_fft,i)+qof(in,i)*iof(taps_fft,i);
}
//calculate inverse FFT on multiplied buffer
fft_execute(plan_inverse);
//add the overlap of the previous segment
complexf* result = plan_inverse->output;
@ -516,35 +516,35 @@ void apply_fir_fft_cc(FFT_PLAN_T* plan, FFT_PLAN_T* plan_inverse, complexf* taps
iof(result,i)/=plan->size;
qof(result,i)/=plan->size;
}
for(int i=0;i<overlap_size;i++) //@apply_fir_fft_cc: add overlap
{
iof(result,i)=iof(result,i)+iof(last_overlap,i);
qof(result,i)=qof(result,i)+qof(last_overlap,i);
}
}
/*
__ __ _ _ _ _
/\ | \/ | | | | | | | | |
/ \ | \ / | __| | ___ _ __ ___ ___ __| |_ _| | __ _| |_ ___ _ __ ___
__ __ _ _ _ _
/\ | \/ | | | | | | | | |
/ \ | \ / | __| | ___ _ __ ___ ___ __| |_ _| | __ _| |_ ___ _ __ ___
/ /\ \ | |\/| | / _` |/ _ \ '_ ` _ \ / _ \ / _` | | | | |/ _` | __/ _ \| '__/ __|
/ ____ \| | | | | (_| | __/ | | | | | (_) | (_| | |_| | | (_| | || (_) | | \__ \
/_/ \_\_| |_| \__,_|\___|_| |_| |_|\___/ \__,_|\__,_|_|\__,_|\__\___/|_| |___/
*/
void amdemod_cf(complexf* input, float *output, int input_size)
{
//@amdemod: i*i+q*q
for (int i=0; i<input_size; i++)
{
{
output[i]=iof(input,i)*iof(input,i)+qof(input,i)*qof(input,i);
}
//@amdemod: sqrt
for (int i=0; i<input_size; i++)
{
{
output[i]=sqrt(output[i]);
}
}
@ -555,7 +555,7 @@ void amdemod_estimator_cf(complexf* input, float *output, int input_size, float
//http://www.dspguru.com/dsp/tricks/magnitude-estimator
//default: optimize for min RMS error
if(alpha==0)
if(alpha==0)
{
alpha=0.947543636291;
beta=0.392485425092;
@ -563,7 +563,7 @@ void amdemod_estimator_cf(complexf* input, float *output, int input_size, float
//@amdemod_estimator
for (int i=0; i<input_size; i++)
{
{
float abs_i=iof(input,i);
if(abs_i<0) abs_i=-abs_i;
float abs_q=qof(input,i);
@ -606,8 +606,8 @@ float fastdcblock_ff(float* input, float* output, int input_size, float last_dc_
avg+=input[i];
}
avg/=input_size;
float avgdiff=avg-last_dc_level;
float avgdiff=avg-last_dc_level;
//DC removal level will change lineraly from last_dc_level to avg.
for(int i=0;i<input_size;i++) //@fastdcblock_ff: remove DC component
{
@ -626,7 +626,7 @@ void fastagc_ff(fastagc_ff_t* input, float* output)
//You have to supply three blocks of samples before the first block comes out.
//AGC reaction speed equals input_size*samp_rate*2
//The algorithm calculates target gain at the end of the first block out of the peak value of all the three blocks.
//The algorithm calculates target gain at the end of the first block out of the peak value of all the three blocks.
//This way the gain change can easily react if there is any peak in the third block.
//Pros: can be easily speeded up with loop vectorization, easy to implement
//Cons: needs 3 buffers, dos not behave similarly to real AGC circuits
@ -643,7 +643,7 @@ void fastagc_ff(fastagc_ff_t* input, float* output)
float target_peak=peak_input;
if(target_peak<input->peak_2) target_peak=input->peak_2;
if(target_peak<input->peak_1) target_peak=input->peak_1;
//we change the gain linearly on the apply_block from the last_gain to target_gain.
float target_gain=input->reference/target_peak;
if(target_gain>FASTAGC_MAX_GAIN) target_gain=FASTAGC_MAX_GAIN;
@ -668,9 +668,9 @@ void fastagc_ff(fastagc_ff_t* input, float* output)
}
/*
______ __ __ _ _ _ _
| ____| \/ | | | | | | | | |
| |__ | \ / | __| | ___ _ __ ___ ___ __| |_ _| | __ _| |_ ___ _ __ ___
______ __ __ _ _ _ _
| ____| \/ | | | | | | | | |
| |__ | \ / | __| | ___ _ __ ___ ___ __| |_ _| | __ _| |_ ___ _ __ ___
| __| | |\/| | / _` |/ _ \ '_ ` _ \ / _ \ / _` | | | | |/ _` | __/ _ \| '__/ __|
| | | | | | | (_| | __/ | | | | | (_) | (_| | |_| | | (_| | || (_) | | \__ \
|_| |_| |_| \__,_|\___|_| |_| |_|\___/ \__,_|\__,_|_|\__,_|\__\___/|_| |___/
@ -721,33 +721,33 @@ complexf fmdemod_quadri_cf(complexf* input, float* output, int input_size, float
temp_dq[0]=qof(input,0)-last_sample.q;
for (int i=1; i<input_size; i++) //@fmdemod_quadri_cf: dq
{
{
temp_dq[i]=qof(input,i)-qof(input,i-1);
}
temp_di[0]=iof(input,0)-last_sample.i;
for (int i=1; i<input_size; i++) //@fmdemod_quadri_cf: di
{
{
temp_di[i]=iof(input,i)-iof(input,i-1);
}
for (int i=0; i<input_size; i++) //@fmdemod_quadri_cf: output numerator
{
{
output[i]=(iof(input,i)*temp_dq[i]-qof(input,i)*temp_di[i]);
}
for (int i=0; i<input_size; i++) //@fmdemod_quadri_cf: output denomiator
{
{
temp[i]=iof(input,i)*iof(input,i)+qof(input,i)*qof(input,i);
}
}
for (int i=0; i<input_size; i++) //@fmdemod_quadri_cf: output division
{
{
output[i]=fmdemod_quadri_K*output[i]/temp[i];
}
return input[input_size-1];
}
inline int is_nan(float f)
inline int is_nan(float f)
{
//http://stackoverflow.com/questions/570669/checking-if-a-double-or-float-is-nan-in-c
unsigned u = *(unsigned*)&f;
@ -757,7 +757,7 @@ inline int is_nan(float f)
float deemphasis_wfm_ff (float* input, float* output, int input_size, float tau, int sample_rate, float last_output)
{
/*
/*
typical time constant (tau) values:
WFM transmission in USA: 75 us -> tau = 75e-6
WFM transmission in EU: 50 us -> tau = 50e-6
@ -767,7 +767,7 @@ float deemphasis_wfm_ff (float* input, float* output, int input_size, float tau,
float dt = 1.0/sample_rate;
float alpha = dt/(tau+dt);
if(is_nan(last_output)) last_output=0.0; //if last_output is NaN
output[0]=alpha*input[0]+(1-alpha)*last_output;
output[0]=alpha*input[0]+(1-alpha)*last_output;
for (int i=1;i<input_size;i++) //@deemphasis_wfm_ff
output[i]=alpha*input[i]+(1-alpha)*output[i-1]; //this is the simplest IIR LPF
return output[input_size-1];
@ -780,7 +780,7 @@ int deemphasis_nfm_ff (float* input, float* output, int input_size, int sample_r
/*
Warning! This only works on predefined samplerates, as it uses fixed FIR coefficients defined in predefined.h
However, there are the octave commands to generate the taps for your custom (fixed) sample rate.
What it does:
What it does:
- reject below 400 Hz
- passband between 400 HZ - 4 kHz, but with 20 dB/decade rolloff (for deemphasis)
- reject everything above 4 kHz
@ -807,10 +807,10 @@ int deemphasis_nfm_ff (float* input, float* output, int input_size, int sample_r
void limit_ff(float* input, float* output, int input_size, float max_amplitude)
{
for (int i=0; i<input_size; i++) //@limit_ff
{
{
output[i]=(max_amplitude<input[i])?max_amplitude:input[i];
output[i]=(-max_amplitude>output[i])?-max_amplitude:output[i];
}
}
}
void gain_ff(float* input, float* output, int input_size, float gain)
@ -818,20 +818,40 @@ void gain_ff(float* input, float* output, int input_size, float gain)
for(int i=0;i<input_size;i++) output[i]=gain*input[i]; //@gain_ff
}
float get_power_f(float* input, int input_size, int decimation)
{
float acc = 0;
for(int i=0;i<input_size;i+=decimation)
{
acc += (input[i]*input[i])/input_size;
}
return acc;
}
float get_power_c(complexf* input, int input_size, int decimation)
{
float acc = 0;
for(int i=0;i<input_size;i+=decimation)
{
acc += (iof(input,i)*iof(input,i)+qof(input,i)*qof(input,i))/input_size;
}
return acc;
}
/*
__ __ _ _ _
| \/ | | | | | | |
| \ / | ___ __| |_ _| | __ _| |_ ___ _ __ ___
__ __ _ _ _
| \/ | | | | | | |
| \ / | ___ __| |_ _| | __ _| |_ ___ _ __ ___
| |\/| |/ _ \ / _` | | | | |/ _` | __/ _ \| '__/ __|
| | | | (_) | (_| | |_| | | (_| | || (_) | | \__ \
|_| |_|\___/ \__,_|\__,_|_|\__,_|\__\___/|_| |___/
*/
void add_dcoffset_cc(complexf* input, complexf* output, int input_size)
{
for(int i=0;i<input_size;i++) iof(output,i)=0.5+iof(input,i)/2;
for(int i=0;i<input_size;i++) qof(output,i)=qof(input,i)/2;
for(int i=0;i<input_size;i++) iof(output,i)=0.5+iof(input,i)/2;
for(int i=0;i<input_size;i++) qof(output,i)=qof(input,i)/2;
}
float fmmod_fc(float* input, complexf* output, int input_size, float last_phase)
@ -854,30 +874,30 @@ void fixed_amplitude_cc(complexf* input, complexf* output, int input_size, float
{
//float phase=atan2(iof(input,i),qof(input,i));
//iof(output,i)=cos(phase)*amp;
//qof(output,i)=sin(phase)*amp;
//qof(output,i)=sin(phase)*amp;
//A faster solution:
float amplitude_now = sqrt(iof(input,i)*iof(input,i)+qof(input,i)*qof(input,i));
float gain = (amplitude_now > 0) ? new_amplitude / amplitude_now : 0;
iof(output,i)=iof(input,i)*gain;
qof(output,i)=qof(input,i)*gain;
}
}
}
/*
______ _ ______ _ _______ __
| ____| | | | ____| (_) |__ __| / _|
| |__ __ _ ___| |_ | |__ ___ _ _ _ __ _ ___ _ __ | |_ __ __ _ _ __ ___| |_ ___ _ __ _ __ ___
| __/ _` / __| __| | __/ _ \| | | | '__| |/ _ \ '__| | | '__/ _` | '_ \/ __| _/ _ \| '__| '_ ` _ \
______ _ ______ _ _______ __
| ____| | | | ____| (_) |__ __| / _|
| |__ __ _ ___| |_ | |__ ___ _ _ _ __ _ ___ _ __ | |_ __ __ _ _ __ ___| |_ ___ _ __ _ __ ___
| __/ _` / __| __| | __/ _ \| | | | '__| |/ _ \ '__| | | '__/ _` | '_ \/ __| _/ _ \| '__| '_ ` _ \
| | | (_| \__ \ |_ | | | (_) | |_| | | | | __/ | | | | | (_| | | | \__ \ || (_) | | | | | | | |
|_| \__,_|___/\__| |_| \___/ \__,_|_| |_|\___|_| |_|_| \__,_|_| |_|___/_| \___/|_| |_| |_| |_|
|_| \__,_|___/\__| |_| \___/ \__,_|_| |_|\___|_| |_|_| \__,_|_| |_|___/_| \___/|_| |_| |_| |_|
*/
int log2n(int x)
{
int result=-1;
for(int i=0;i<31;i++)
for(int i=0;i<31;i++)
{
if((x>>i)&1) //@@log2n
{
@ -892,7 +912,7 @@ int next_pow2(int x)
{
int pow2;
//portability? (31 is the problem)
for(int i=0;i<31;i++)
for(int i=0;i<31;i++)
{
if(x<(pow2=1<<i)) return pow2; //@@next_pow2
}
@ -923,21 +943,21 @@ void apply_window_f(float* input, float* output, int size, window_t window)
void logpower_cf(complexf* input, float* output, int size, float add_db)
{
for(int i=0;i<size;i++) output[i]=iof(input,i)*iof(input,i) + qof(input,i)*qof(input,i); //@logpower_cf: pass 1
for(int i=0;i<size;i++) output[i]=log10(output[i]); //@logpower_cf: pass 2
for(int i=0;i<size;i++) output[i]=10*output[i]+add_db; //@logpower_cf: pass 3
}
/*
_____ _ _
| __ \ | | (_)
| | | | __ _| |_ __ _ ___ ___ _ ____ _____ _ __ ___ _ ___ _ __
| | | |/ _` | __/ _` | / __/ _ \| '_ \ \ / / _ \ '__/ __| |/ _ \| '_ \
_____ _ _
| __ \ | | (_)
| | | | __ _| |_ __ _ ___ ___ _ ____ _____ _ __ ___ _ ___ _ __
| | | |/ _` | __/ _` | / __/ _ \| '_ \ \ / / _ \ '__/ __| |/ _ \| '_ \
| |__| | (_| | || (_| | | (_| (_) | | | \ V / __/ | \__ \ | (_) | | | |
|_____/ \__,_|\__\__,_| \___\___/|_| |_|\_/ \___|_| |___/_|\___/|_| |_|
*/
*/
void convert_u8_f(unsigned char* input, float* output, int input_size)
{
@ -957,7 +977,7 @@ void convert_s16_f(short* input, float* output, int input_size)
void convert_f_u8(float* input, unsigned char* output, int input_size)
{
for(int i=0;i<input_size;i++) output[i]=input[i]*UCHAR_MAX*0.5+128; //@convert_f_u8
//128 above is the correct value to add. In any other case a DC component
//128 above is the correct value to add. In any other case a DC component
//of at least -60 dB is shown on the FFT plot after convert_f_u8 -> convert_u8_f
}
@ -968,7 +988,7 @@ void convert_f_s8(float* input, signed char* output, int input_size)
void convert_f_s16(float* input, short* output, int input_size)
{
/*for(int i=0;i<input_size;i++)
/*for(int i=0;i<input_size;i++)
{
if(input[i]>1.0) input[i]=1.0;
if(input[i]<-1.0) input[i]=-1.0;
@ -989,5 +1009,3 @@ int trivial_vectorize()
}
return c[0];
}

26
libcsdr.h
View file

@ -1,5 +1,5 @@
/*
This software is part of libcsdr, a set of simple DSP routines for
This software is part of libcsdr, a set of simple DSP routines for
Software Defined Radio.
Copyright (c) 2014, Andras Retzler <randras@sdr.hu>
@ -32,14 +32,14 @@ SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#define MIN_M(x,y) (((x)>(y))?(y):(x))
/*
_____ _
/ ____| | |
| | ___ _ __ ___ _ __ | | _____ __
| | / _ \| '_ ` _ \| '_ \| |/ _ \ \/ /
| |___| (_) | | | | | | |_) | | __/> <
\_____\___/|_| |_| |_| .__/|_|\___/_/\_\
| |
|_|
_____ _
/ ____| | |
| | ___ _ __ ___ _ __ | | _____ __
| | / _ \| '_ ` _ \| '_ \| |/ _ \ \/ /
| |___| (_) | | | | | | |_) | | __/> <
\_____\___/|_| |_| |_| .__/|_|\___/_/\_\
| |
|_|
*/
typedef struct complexf_s { float i; float q; } complexf;
@ -64,7 +64,7 @@ typedef struct complexf_s { float i; float q; } complexf;
#define PI ((float)3.14159265358979323846)
//window
typedef enum window_s
typedef enum window_s
{
WINDOW_BOXCAR, WINDOW_BLACKMAN, WINDOW_HAMMING
} window_t;
@ -113,7 +113,7 @@ float fastdcblock_ff(float* input, float* output, int input_size, float last_dc_
typedef struct fastagc_ff_s
{
float* buffer_1;
float* buffer_1;
float* buffer_2;
float* buffer_input; //it is the actual input buffer to fill
float peak_1;
@ -150,7 +150,7 @@ fractional_decimator_ff_t fractional_decimator_ff(float* input, float* output, i
typedef struct shift_table_data_s
{
float* table;
int table_size;
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);
@ -161,6 +161,8 @@ int log2n(int x);
int next_pow2(int x);
void apply_fir_fft_cc(FFT_PLAN_T* plan, FFT_PLAN_T* plan_inverse, complexf* taps_fft, complexf* last_overlap, int overlap_size);
void gain_ff(float* input, float* output, int input_size, float gain);
float get_power_f(float* input, int input_size, int decimation);
float get_power_c(complexf* input, int input_size, int decimation);
void add_dcoffset_cc(complexf* input, complexf* output, int input_size);
float fmmod_fc(float* input, complexf* output, int input_size, float last_phase);