diff --git a/Readme.md b/Readme.md index 082d8a08c..3a3468023 100644 --- a/Readme.md +++ b/Readme.md @@ -105,14 +105,15 @@ Done since the fork - New plugin for BladeRF interfacing libbladeRF directly - Corrected the nasty audio band resampling bug preventing use of sample rates that are not power of 2 multiples of 48kHz. This was because the resampling ratio was calculated with an integer division instead of a float division. - As a consequence of the above added more interesting values for the available sampling rates of the BladeRF plugin + - Variable span for the SSB demod down to 1.5 kHz + - Filter out CTCSS tones in NFMDemod ===== To Do ===== - Enhance WFM (stereo, RDS?) - - Possibility to completely undock the receiver in a separate window. Useful when there are many receivers - - Even larger decimation capability for narrowband and very narrowband work? (64, ...) - Even more demods ... - Triggering capability like on expensive spectrum analyzers to trap burst signals - recording capability + - Tx channels for Rx/Tx boards like BladeRF diff --git a/include-gpl/dsp/bandpass.h b/include-gpl/dsp/bandpass.h new file mode 100644 index 000000000..fb632079c --- /dev/null +++ b/include-gpl/dsp/bandpass.h @@ -0,0 +1,120 @@ +#ifndef INCLUDE_BANDPASS_H +#define INCLUDE_BANDPASS_H + +#define _USE_MATH_DEFINES +#include +#include "dsp/dsptypes.h" + +template class Bandpass { +public: + Bandpass() { } + + void create(int nTaps, double sampleRate, double lowCutoff, double highCutoff) + { + std::vector taps_lp; + std::vector taps_hp; + double wcl = 2.0 * M_PI * lowCutoff; + double Wcl = wcl / sampleRate; + double wch = 2.0 * M_PI * highCutoff; + double Wch = wch / sampleRate; + int i; + + // check constraints + if(!(nTaps & 1)) { + qDebug("Bandpass filter has to have an odd number of taps"); + nTaps++; + } + + // make room + m_samples.resize(nTaps); + for(int i = 0; i < nTaps; i++) + m_samples[i] = 0; + m_ptr = 0; + m_taps.resize(nTaps / 2 + 1); + taps_lp.resize(nTaps / 2 + 1); + taps_hp.resize(nTaps / 2 + 1); + + // generate Sinc filter core + for(i = 0; i < nTaps / 2 + 1; i++) { + if(i == (nTaps - 1) / 2) { + taps_lp[i] = Wch / M_PI; + taps_hp[i] = -(Wcl / M_PI); + } + else { + taps_lp[i] = sin(((double)i - ((double)nTaps - 1.0) / 2.0) * Wch) / (((double)i - ((double)nTaps - 1.0) / 2.0) * M_PI); + taps_hp[i] = -sin(((double)i - ((double)nTaps - 1.0) / 2.0) * Wcl) / (((double)i - ((double)nTaps - 1.0) / 2.0) * M_PI); + } + } + + taps_hp[(nTaps - 1) / 2] += 1; + + // apply Hamming window and combine lowpass and highpass + for(i = 0; i < nTaps / 2 + 1; i++) { + taps_lp[i] *= 0.54 + 0.46 * cos((2.0 * M_PI * ((double)i - ((double)nTaps - 1.0) / 2.0)) / (double)nTaps); + taps_hp[i] *= 0.54 + 0.46 * cos((2.0 * M_PI * ((double)i - ((double)nTaps - 1.0) / 2.0)) / (double)nTaps); + m_taps[i] = -(taps_lp[i]+taps_hp[i]); + } + + m_taps[(nTaps - 1) / 2] += 1; + + // normalize + Real sum = 0; + + for(i = 0; i < (int)m_taps.size() - 1; i++) { + sum += m_taps[i] * 2; + } + + sum += m_taps[i]; + + for(i = 0; i < (int)m_taps.size(); i++) { + m_taps[i] /= sum; + } + } + + Type filter(Type sample) + { + Type acc = 0; + int a = m_ptr; + int b = a - 1; + int i; + + m_samples[m_ptr] = sample; + + while(b < 0) { + b += m_samples.size(); + } + + for(i = 0; i < (int)m_taps.size() - 1; i++) + { + acc += (m_samples[a] + m_samples[b]) * m_taps[i]; + a++; + + while(a >= (int)m_samples.size()) { + a -= m_samples.size(); + } + + b--; + + while(b < 0) { + b += m_samples.size(); + } + } + + acc += m_samples[a] * m_taps[i]; + + m_ptr++; + + while(m_ptr >= (int)m_samples.size()) { + m_ptr -= m_samples.size(); + } + + return acc; + } + +private: + std::vector m_taps; + std::vector m_samples; + int m_ptr; +}; + +#endif // INCLUDE_BANDPASS_H diff --git a/include-gpl/dsp/highpass.h b/include-gpl/dsp/highpass.h new file mode 100644 index 000000000..407cdc1db --- /dev/null +++ b/include-gpl/dsp/highpass.h @@ -0,0 +1,90 @@ +#ifndef INCLUDE_HIGHPASS_H +#define INCLUDE_HIGHPASS_H + +#define _USE_MATH_DEFINES +#include +#include "dsp/dsptypes.h" + +template class Highpass { +public: + Highpass() { } + + void create(int nTaps, double sampleRate, double cutoff) + { + double wc = 2.0 * M_PI * cutoff; + double Wc = wc / sampleRate; + int i; + + // check constraints + if(!(nTaps & 1)) { + qDebug("Highpass filter has to have an odd number of taps"); + nTaps++; + } + + // make room + m_samples.resize(nTaps); + for(int i = 0; i < nTaps; i++) + m_samples[i] = 0; + m_ptr = 0; + m_taps.resize(nTaps / 2 + 1); + + // generate Sinc filter core for lowpass but inverting every other tap for highpass keeping center tap + for(i = 0; i < nTaps / 2 + 1; i++) { + if(i == (nTaps - 1) / 2) + m_taps[i] = -(Wc / M_PI); + else + m_taps[i] = -sin(((double)i - ((double)nTaps - 1.0) / 2.0) * Wc) / (((double)i - ((double)nTaps - 1.0) / 2.0) * M_PI); + } + + m_taps[(nTaps - 1) / 2] += 1; + + // apply Hamming window + for(i = 0; i < nTaps / 2 + 1; i++) + m_taps[i] *= 0.54 + 0.46 * cos((2.0 * M_PI * ((double)i - ((double)nTaps - 1.0) / 2.0)) / (double)nTaps); + + // normalize + Real sum = 0; + for(i = 0; i < (int)m_taps.size() - 1; i++) + sum += m_taps[i] * 2; + sum += m_taps[i]; + for(i = 0; i < (int)m_taps.size(); i++) + m_taps[i] /= sum; + } + + Type filter(Type sample) + { + Type acc = 0; + int a = m_ptr; + int b = a - 1; + int i; + + m_samples[m_ptr] = sample; + + while(b < 0) + b += m_samples.size(); + + for(i = 0; i < (int)m_taps.size() - 1; i++) { + acc += (m_samples[a] + m_samples[b]) * m_taps[i]; + a++; + while(a >= (int)m_samples.size()) + a -= m_samples.size(); + b--; + while(b < 0) + b += m_samples.size(); + } + acc += m_samples[a] * m_taps[i]; + + m_ptr++; + while(m_ptr >= (int)m_samples.size()) + m_ptr -= m_samples.size(); + + return acc; + } + +private: + std::vector m_taps; + std::vector m_samples; + int m_ptr; +}; + +#endif // INCLUDE_HIGHPASS_H diff --git a/plugins/channel/nfm/nfmdemod.cpp b/plugins/channel/nfm/nfmdemod.cpp index 9432df232..0334691c5 100644 --- a/plugins/channel/nfm/nfmdemod.cpp +++ b/plugins/channel/nfm/nfmdemod.cpp @@ -147,17 +147,11 @@ void NFMDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iter // AF processing - demod = m_lowpass.filter(demod); - - /* - if(demod < -1) - demod = -1; - else if(demod > 1) - demod = 1; - */ - + //demod = m_lowpass.filter(demod); + demod = m_bandpass.filter(demod); demod *= m_running.m_volume; - sample = demod * 32700; + //sample = demod * 32700; + sample = demod * ((1<<16)/301); // denominator = bandpass filter number of taps } else { m_AGC.close(); @@ -245,7 +239,8 @@ void NFMDemod::apply() if((m_config.m_afBandwidth != m_running.m_afBandwidth) || (m_config.m_audioSampleRate != m_running.m_audioSampleRate)) { - m_lowpass.create(21, m_config.m_audioSampleRate, m_config.m_afBandwidth); + //m_lowpass.create(21, m_config.m_audioSampleRate, m_config.m_afBandwidth); + m_bandpass.create(301, m_config.m_audioSampleRate, 300.0, m_config.m_afBandwidth); } if(m_config.m_squelch != m_running.m_squelch) { diff --git a/plugins/channel/nfm/nfmdemod.h b/plugins/channel/nfm/nfmdemod.h index 9996f8bcd..1bb0830cc 100644 --- a/plugins/channel/nfm/nfmdemod.h +++ b/plugins/channel/nfm/nfmdemod.h @@ -23,6 +23,7 @@ #include "dsp/nco.h" #include "dsp/interpolator.h" #include "dsp/lowpass.h" +#include "dsp/bandpass.h" #include "dsp/movingaverage.h" #include "dsp/agc.h" #include "audio/audiofifo.h" @@ -111,7 +112,8 @@ private: Interpolator m_interpolator; Real m_interpolatorDistance; Real m_interpolatorDistanceRemain; - Lowpass m_lowpass; + //Lowpass m_lowpass; + Bandpass m_bandpass; Real m_squelchLevel; int m_squelchState;