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UHSDR/UHSDR-active-devel/mchf-eclipse/drivers/audio/audio_driver.c
Xavier f838fa66c4 first commit
2022-11-08 16:13:55 +01:00

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/* -*- mode: c; tab-width: 4; indent-tabs-mode: t; c-basic-offset: 4; coding: utf-8 -*- */
/************************************************************************************
** **
** mcHF QRP Transceiver **
** K Atanassov - M0NKA 2014 **
** **
**---------------------------------------------------------------------------------**
** **
** File name: **
** Description: **
** Last Modified: **
** Licence: GNU GPLv3 **
************************************************************************************/
// Common
#include <assert.h>
#include "uhsdr_board.h"
#include "ui_driver.h"
#include "profiling.h"
#include <stdio.h>
#include <math.h>
#include "codec.h"
#include "cw_gen.h"
#include <limits.h>
#include "softdds.h"
#include "audio_driver.h"
#include "audio_nr.h"
#include "audio_agc.h"
#include "audio_management.h"
#include "radio_management.h"
#include "usbd_audio_if.h"
#include "ui_spectrum.h"
#include "filters.h"
#include "uhsdr_hw_i2s.h"
#include "rtty.h"
#include "psk.h"
#include "cw_decoder.h"
#include "freedv_uhsdr.h"
#include "freq_shift.h"
#include "audio_nr.h"
#ifdef USE_CONVOLUTION
#include "audio_convolution.h"
#endif
#include "fm_subaudible_tone_table.h" // hm.
#include "uhsdr_math.h"
#include "tx_processor.h"
#include "audio_reverb.h"
// SSB filters - now handled in ui_driver to allow I/Q phase adjustment
#define LMS2_NOTCH_STATE_ARRAY_SIZE (DSP_NOTCH_NUMTAPS_MAX + IQ_BLOCK_SIZE)
#ifdef USE_LMS_AUTONOTCH
typedef struct
{
float32_t errsig2[IQ_BLOCK_SIZE];
arm_lms_norm_instance_f32 lms2Norm_instance;
arm_lms_instance_f32 lms2_instance;
float32_t lms2StateF32[LMS2_NOTCH_STATE_ARRAY_SIZE];
float32_t lms2NormCoeff_f32[DSP_NOTCH_NUMTAPS_MAX];
float32_t lms2_nr_delay[DSP_NOTCH_BUFLEN_MAX];
} LMSData;
#endif
// Decimator for Zoom FFT
static arm_fir_decimate_instance_f32 DECIMATE_ZOOM_FFT_I;
float32_t __MCHF_SPECIALMEM decimZoomFFTIState[FIR_RXAUDIO_BLOCK_SIZE + FIR_RXAUDIO_NUM_TAPS];
// Decimator for Zoom FFT
static arm_fir_decimate_instance_f32 DECIMATE_ZOOM_FFT_Q;
float32_t __MCHF_SPECIALMEM decimZoomFFTQState[FIR_RXAUDIO_BLOCK_SIZE + FIR_RXAUDIO_NUM_TAPS];
// Audio RX - Interpolator
static arm_fir_interpolate_instance_f32 INTERPOLATE_RX[NUM_AUDIO_CHANNELS];
float32_t __MCHF_SPECIALMEM interpState[NUM_AUDIO_CHANNELS][FIR_RXAUDIO_BLOCK_SIZE + FIR_RXAUDIO_NUM_TAPS];
#define NR_INTERPOLATE_NO_TAPS 40
static arm_fir_decimate_instance_f32 DECIMATE_NR;
float32_t decimNRState[FIR_RXAUDIO_BLOCK_SIZE + 4];
static arm_fir_interpolate_instance_f32 INTERPOLATE_NR;
float32_t interplNRState[FIR_RXAUDIO_BLOCK_SIZE + NR_INTERPOLATE_NO_TAPS];
#define IIR_RX_STATE_ARRAY_SIZE (IIR_RXAUDIO_BLOCK_SIZE + IIR_RXAUDIO_NUM_STAGES_MAX)
// variables for RX IIR filters
static float32_t iir_rx_state[NUM_AUDIO_CHANNELS][IIR_RX_STATE_ARRAY_SIZE];
static arm_iir_lattice_instance_f32 IIR_PreFilter[NUM_AUDIO_CHANNELS];
// variables for RX antialias IIR filter
static float32_t iir_aa_state[NUM_AUDIO_CHANNELS][IIR_RX_STATE_ARRAY_SIZE];
static arm_iir_lattice_instance_f32 IIR_AntiAlias[NUM_AUDIO_CHANNELS];
// static float32_t Koeff[20];
// variables for RX manual notch, manual peak & bass shelf IIR biquad filter
static arm_biquad_casd_df1_inst_f32 IIR_biquad_1[NUM_AUDIO_CHANNELS] =
{
{
.numStages = 4,
.pCoeffs = (float32_t *)(float32_t [])
{
1,0,0,0,0, 1,0,0,0,0, 1,0,0,0,0, 1,0,0,0,0
}, // 4 x 5 = 20 coefficients
.pState = (float32_t *)(float32_t [])
{
0,0,0,0, 0,0,0,0, 0,0,0,0, 0,0,0,0
} // 4 x 4 = 16 state variables
},
#ifdef USE_TWO_CHANNEL_AUDIO
{
.numStages = 4,
.pCoeffs = (float32_t *)(float32_t [])
{
1,0,0,0,0, 1,0,0,0,0, 1,0,0,0,0, 1,0,0,0,0
}, // 3 x 5 = 15 coefficients
.pState = (float32_t *)(float32_t [])
{
0,0,0,0, 0,0,0,0, 0,0,0,0, 0,0,0,0
} // 3 x 4 = 12 state variables
}
#endif
};
// variables for RX treble shelf IIR biquad filter
static arm_biquad_casd_df1_inst_f32 IIR_biquad_2[NUM_AUDIO_CHANNELS] =
{
{
.numStages = 1,
.pCoeffs = (float32_t *)(float32_t [])
{
1,0,0,0,0
}, // 1 x 5 = 5 coefficients
.pState = (float32_t *)(float32_t [])
{
0,0,0,0
} // 1 x 4 = 4 state variables
},
#ifdef USE_TWO_CHANNEL_AUDIO
{
.numStages = 1,
.pCoeffs = (float32_t *)(float32_t [])
{
1,0,0,0,0
}, // 1 x 5 = 5 coefficients
.pState = (float32_t *)(float32_t [])
{
0,0,0,0
} // 1 x 4 = 4 state variables
}
#endif
};
// variables for ZoomFFT lowpass filtering
static arm_biquad_casd_df1_inst_f32 IIR_biquad_Zoom_FFT_I =
{
.numStages = 4,
.pCoeffs = (float32_t *)(float32_t [])
{
1,0,0,0,0, 1,0,0,0,0 // passthru
}, // 2 x 5 = 10 coefficients
.pState = (float32_t *)(float32_t [])
{
0,0,0,0, 0,0,0,0, 0,0,0,0, 0,0,0,0
} // 4 x 4 = 16 state variables
};
static arm_biquad_casd_df1_inst_f32 IIR_biquad_Zoom_FFT_Q =
{
.numStages = 4,
.pCoeffs = (float32_t *)(float32_t [])
{
1,0,0,0,0, 1,0,0,0,0 // passthru
}, // 2 x 5 = 10 coefficients
.pState = (float32_t *)(float32_t [])
{
0,0,0,0, 0,0,0,0, 0,0,0,0, 0,0,0,0
} // 4 x 4 = 16 state variables
};
// sr = 12ksps, Fstop = 2k7, we lowpass-filtered the audio already in the main aido path (IIR),
// so only the minimum size filter (4 taps) is used here
static float32_t NR_decimate_coeffs [4] = {0.099144206287089282, 0.492752007869707798, 0.492752007869707798, 0.099144206287089282};
// 12ksps, Fstop = 2k7, KAISER, 40 taps, a good interpolation filter after the interpolation, maybe too many taps?
static float32_t NR_interpolate_coeffs [NR_INTERPOLATE_NO_TAPS] = {-495.0757586677611930E-6, 0.001320676868426568, 0.001533845835568487,-0.002357633129357554,-0.003572560455091757, 0.003388797052024843, 0.007032840952358404,-0.003960820803871866,-0.012365795129023015, 0.003357357660531775, 0.020101326014980946,-475.4964584295063900E-6,-0.031094910247864812,-0.006597041050034579, 0.047436525317202147, 0.022324808965607446,-0.076541709512474090,-0.064246467306504046, 0.167750545742874818, 0.427794841657261171, 0.427794841657261171, 0.167750545742874818,-0.064246467306504046,-0.076541709512474090, 0.022324808965607446, 0.047436525317202147,-0.006597041050034579,-0.031094910247864812,-475.4964584295063900E-6, 0.020101326014980946, 0.003357357660531775,-0.012365795129023015,-0.003960820803871866, 0.007032840952358404, 0.003388797052024843,-0.003572560455091757,-0.002357633129357554, 0.001533845835568487, 0.001320676868426568,-495.0757586677611930E-6};
// this is wrong! Interpolation filters act at the sample rate AFTER the interpolation, in this case at 12ksps
// 6ksps, Fstop = 2k65, KAISER
//static float32_t NR_interpolate_coeffs [NR_INTERPOLATE_NO_TAPS] = {-903.6623076669911820E-6, 0.001594488333496738,-0.002320508982899863, 0.002832351511451895,-0.002797105957386612, 0.001852836963547170, 308.6133633078010230E-6,-0.003842008360761881, 0.008649943961959465,-0.014305251526745446, 0.020012524686320185,-0.024618364878703208, 0.026664997481476788,-0.024458388333600374, 0.016080841021827566, 818.1032282579135430E-6,-0.029933800539235892, 0.079833661336890141,-0.182038248016552551, 0.626273078268197225, 0.626273078268197225,-0.182038248016552551, 0.079833661336890141,-0.029933800539235892, 818.1032282579135430E-6, 0.016080841021827566,-0.024458388333600374, 0.026664997481476788,-0.024618364878703208, 0.020012524686320185,-0.014305251526745446, 0.008649943961959465,-0.003842008360761881, 308.6133633078010230E-6, 0.001852836963547170,-0.002797105957386612, 0.002832351511451895,-0.002320508982899863, 0.001594488333496738,-903.6623076669911820E-6};
static float32_t* mag_coeffs[MAGNIFY_NUM] =
{
// for Index 0 [1xZoom == no zoom] the mag_coeffs will a NULL ptr, since the filter is not going to be used in this mode!
NULL,
(float32_t*)(const float32_t[]) {
// 2x magnify - index 1
// 12kHz, sample rate 48k, 60dB stopband, elliptic
// a1 and coeffs[A2] negated! order: coeffs[B0], coeffs[B1], coeffs[B2], a1, coeffs[A2]
// Iowa Hills IIR Filter Designer, DD4WH Aug 16th 2016
0.228454526413293696,
0.077639329099949764,
0.228454526413293696,
0.635534925142242080,
-0.170083307068779194,
0.436788292542003964,
0.232307972937606161,
0.436788292542003964,
0.365885230717786780,
-0.471769788739400842,
0.535974654742658707,
0.557035600464780845,
0.535974654742658707,
0.125740787233286133,
-0.754725697183384336,
0.501116342273565607,
0.914877831284765408,
0.501116342273565607,
0.013862536615004284,
-0.930973052446900984
},
(float32_t*)(const float32_t[])
{
// 4x magnify - index 2
// 6kHz, sample rate 48k, 60dB stopband, elliptic
// a1 and coeffs[A2] negated! order: coeffs[B0], coeffs[B1], coeffs[B2], a1, coeffs[A2]
// Iowa Hills IIR Filter Designer, DD4WH Aug 16th 2016
0.182208761527446556,
-0.222492493114674145,
0.182208761527446556,
1.326111070880959810,
-0.468036100821178802,
0.337123762652097259,
-0.366352718812586853,
0.337123762652097259,
1.337053579516321200,
-0.644948386007929031,
0.336163175380826074,
-0.199246162162897811,
0.336163175380826074,
1.354952684569386670,
-0.828032873168141115,
0.178588201750411041,
0.207271695028067304,
0.178588201750411041,
1.386486967455699220,
-0.950935065984588657
},
(float32_t*)(const float32_t[])
{
// 8x magnify - index 3
// 3kHz, sample rate 48k, 60dB stopband, elliptic
// a1 and coeffs[A2] negated! order: coeffs[B0], coeffs[B1], coeffs[B2], a1, coeffs[A2]
// Iowa Hills IIR Filter Designer, DD4WH Aug 16th 2016
0.185643392652478922,
-0.332064345389014803,
0.185643392652478922,
1.654637402827731090,
-0.693859842743674182,
0.327519300813245984,
-0.571358085216950418,
0.327519300813245984,
1.715375037176782860,
-0.799055553586324407,
0.283656142708241688,
-0.441088976843048652,
0.283656142708241688,
1.778230635987093860,
-0.904453944560528522,
0.079685368654848945,
-0.011231810140649204,
0.079685368654848945,
1.825046003243238070,
-0.973184930412286708
},
(float32_t*)(const float32_t[])
{
// 16x magnify - index 4
// 1k5, sample rate 48k, 60dB stopband, elliptic
// a1 and coeffs[A2] negated! order: coeffs[B0], coeffs[B1], coeffs[B2], a1, coeffs[A2]
// Iowa Hills IIR Filter Designer, DD4WH Aug 16th 2016
0.194769868656866380,
-0.379098413160710079,
0.194769868656866380,
1.824436402073870810,
-0.834877726226893380,
0.333973874901496770,
-0.646106479315673776,
0.333973874901496770,
1.871892825636887640,
-0.893734096124207178,
0.272903880596429671,
-0.513507745397738469,
0.272903880596429671,
1.918161772571113750,
-0.950461788366234739,
0.053535383722369843,
-0.069683422367188122,
0.053535383722369843,
1.948900719896301760,
-0.986288064973853129
},
(float32_t*)(const float32_t[])
{
// 32x magnify - index 5
// 750Hz, sample rate 48k, 60dB stopband, elliptic
// a1 and coeffs[A2] negated! order: coeffs[B0], coeffs[B1], coeffs[B2], a1, coeffs[A2]
// Iowa Hills IIR Filter Designer, DD4WH Aug 16th 2016
0.201507402588557594,
-0.400273615727755550,
0.201507402588557594,
1.910767558906650840,
-0.913508748356010480,
0.340295203367131205,
-0.674930558961690075,
0.340295203367131205,
1.939398230905991390,
-0.945058078678563840,
0.271859921641011359,
-0.535453706265515361,
0.271859921641011359,
1.966439529620203740,
-0.974705666636711099,
0.047026497485465592,
-0.084562104085501480,
0.047026497485465592,
1.983564238653704900,
-0.993055129539134551
}
};
#ifdef USE_SIMPLE_FREEDV_FILTERS
//******* From here 2 set of filters for the I/Q FreeDV aliasing filter**********
// I- and Q- Filter instances for FreeDV downsampling aliasing filters
static arm_biquad_casd_df1_inst_f32 IIR_biquad_FreeDV_I =
{
.numStages = 2,
.pCoeffs = (float32_t *)(float32_t [])
{
1,0,0,0,0, 1,0,0,0,0 // passthru
}, // 2 x 5 = 10 coefficients
.pState = (float32_t *)(float32_t [])
{
0,0,0,0, 0,0,0,0
} // 2 x 4 = 8 state variables
};
static arm_biquad_casd_df1_inst_f32 IIR_biquad_FreeDV_Q =
{
.numStages = 2,
.pCoeffs = (float32_t *)(float32_t [])
{
1,0,0,0,0, 1,0,0,0,0 // passthru
}, // 2 x 5 = 10 coefficients
.pState = (float32_t *)(float32_t [])
{
0,0,0,0, 0,0,0,0
} // 2 x 4 = 8 state variables
};
static float32_t* FreeDV_coeffs[1] =
{
(float32_t*)(const float32_t[]){
// index 1
// 2,4kHz, sample rate 48k, 50dB stopband, elliptic
// only 2 stages!!!!!
// a1 and coeffs[A2] negated! order: coeffs[B0], coeffs[B1], coeffs[B2], a1, coeffs[A2]
// Iowa Hills IIR Filter Designer, DL2FW 20-10-16
0.083165011486267731,
-0.118387356334666696,
0.083165011486267731,
1.666027486884941840,
-0.713970153522810569,
0.068193683664968877,
-0.007220581127135660,
0.068193683664968877,
1.763363677375461290,
-0.892530463578263378
}
};
#endif
//******* End of 2 set of filters for the I/Q FreeDV aliasing filter**********
// S meter public
SMeter sm;
// ATTENTION: These data structures have been placed in CCM Memory (64k)
// IF THE SIZE OF THE DATA STRUCTURE GROWS IT WILL QUICKLY BE OUT OF SPACE IN CCM
// Be careful! Check mchf-eclipse.map for current allocation
AudioDriverState __MCHF_SPECIALMEM ads;
AudioDriverBuffer __MCHF_SPECIALMEM adb;
#if defined(USE_LMS_AUTONOTCH)
LMSData __MCHF_SPECIALMEM lmsData;
#endif
#ifdef USE_LEAKY_LMS
lLMS leakyLMS;
#endif
SnapCarrier sc;
/**
* @returns offset frequency in Hz for current frequency translate mode
*/
int32_t AudioDriver_GetTranslateFreq()
{
int32_t fdelta = 0;
switch (ts.iq_freq_mode)
{
case FREQ_IQ_CONV_P6KHZ:
fdelta = 6000;
break;
case FREQ_IQ_CONV_M6KHZ:
fdelta = -6000;
break;
case FREQ_IQ_CONV_P12KHZ:
fdelta = 12000;
break;
case FREQ_IQ_CONV_M12KHZ:
fdelta = -12000;
break;
case FREQ_IQ_CONV_SLIDE:
fdelta = ts.iq_freq_delta;
break;
}
return fdelta;
}
// RX variables for FM squelch IIR filters
static float32_t iir_squelch_rx_state[IIR_RX_STATE_ARRAY_SIZE];
static arm_iir_lattice_instance_f32 IIR_Squelch_HPF;
static void AudioDriver_FM_Rx_Init(fm_conf_t* fm)
{
// RX
fm->sql_avg = 0; // init FM squelch averaging
fm->squelched = true; // TRUE if FM receiver audio is to be squelched, we start squelched.
fm->subaudible_tone_detected = false; // TRUE if subaudible tone has been detected
AudioManagement_CalcSubaudibleDetFreq(fm_subaudible_tone_table[ts.fm_subaudible_tone_det_select]); // RX load/set current FM subaudible tone settings for detection
// Initialize high-pass filter used for the FM noise squelch
IIR_Squelch_HPF.pkCoeffs = IIR_15k_hpf.pkCoeffs; // point to reflection coefficients
IIR_Squelch_HPF.pvCoeffs = IIR_15k_hpf.pvCoeffs; // point to ladder coefficients
IIR_Squelch_HPF.numStages = IIR_15k_hpf.numStages; // number of stages
IIR_Squelch_HPF.pState = iir_squelch_rx_state; // point to state array for IIR filter
arm_fill_f32(0.0,iir_squelch_rx_state,IIR_RX_STATE_ARRAY_SIZE);
}
#ifdef USE_LEAKY_LMS
static void AudioDriver_LeakyLmsNr_Init(void) // LEAKY LMS noise reduction variables
{
leakyLMS.n_taps = 64; //64;
leakyLMS.delay = 16; //16;
leakyLMS.dline_size = LEAKYLMSDLINE_SIZE;
leakyLMS.position = 0;
leakyLMS.two_mu = 0.0001;
leakyLMS.two_mu_int = 100;
leakyLMS.gamma = 0.1; // gamma --> "leakage"
leakyLMS.gamma_int = 100;
leakyLMS.lidx = 120.0;
leakyLMS.lidx_min = 0.0;
leakyLMS.lidx_max = 200.0;
leakyLMS.ngamma = 0.001;
leakyLMS.den_mult = 6.25e-10;
leakyLMS.lincr = 1.0;
leakyLMS.ldecr = 3.0;
leakyLMS.mask = LEAKYLMSDLINE_SIZE - 1;
leakyLMS.in_idx = 0;
leakyLMS.on = 0;
leakyLMS.notch = 0;
// from config
leakyLMS.n_taps = ts.anr_n_taps;
leakyLMS.delay = ts.anr_delay;
leakyLMS.two_mu_int = ts.anr_two_mu_int;
leakyLMS.two_mu = leakyLMS.two_mu_int / 1000000.0;
leakyLMS.gamma_int = ts.anr_gamma_int;
leakyLMS.gamma = leakyLMS.gamma_int / 1000.0;
}
// Automatic noise reduction
// Variable-leak LMS algorithm
// taken from (c) Warren Pratts wdsp library 2016
// GPLv3 licensed
void AudioDriver_LeakyLmsNr (float32_t *in_buff, float32_t *out_buff, int buff_size, bool notch)
{
uint8_t i, j, idx;
float32_t y, error, sigma, inv_sigp, c0, c1, nel, nev;
for (i = 0; i < buff_size; i++)
{
leakyLMS.d[leakyLMS.in_idx] = in_buff[i];
y = 0;
sigma = 0;
for (j = 0; j < leakyLMS.n_taps; j++)
{
idx = (leakyLMS.in_idx + j + leakyLMS.delay) & leakyLMS.mask;
y += leakyLMS.w[j] * leakyLMS.d[idx];
sigma += leakyLMS.d[idx] * leakyLMS.d[idx];
}
inv_sigp = 1.0 / (sigma + 1e-10);
error = leakyLMS.d[leakyLMS.in_idx] - y;
if(notch)
{ // automatic notch filter
out_buff[i] = error;
}
else
{ // noise reduction
// out_buff[i] = y;
out_buff[i] = y * 2; // volume up
}
if((nel = error * (1.0 - leakyLMS.two_mu * sigma * inv_sigp)) < 0.0) nel = -nel;
if((nev = leakyLMS.d[leakyLMS.in_idx] - (1.0 - leakyLMS.two_mu * leakyLMS.ngamma) * y - leakyLMS.two_mu * error * sigma * inv_sigp) < 0.0) nev = -nev;
if (nev < nel)
{
if((leakyLMS.lidx += leakyLMS.lincr) > leakyLMS.lidx_max) leakyLMS.lidx = leakyLMS.lidx_max;
}
else
{
if((leakyLMS.lidx -= leakyLMS.ldecr) < leakyLMS.lidx_min) leakyLMS.lidx = leakyLMS.lidx_min;
}
leakyLMS.ngamma = leakyLMS.gamma * (leakyLMS.lidx * leakyLMS.lidx) * (leakyLMS.lidx * leakyLMS.lidx) * leakyLMS.den_mult;
c0 = 1.0 - leakyLMS.two_mu * leakyLMS.ngamma;
c1 = leakyLMS.two_mu * error * inv_sigp;
for (j = 0; j < leakyLMS.n_taps; j++)
{
idx = (leakyLMS.in_idx + j + leakyLMS.delay) & leakyLMS.mask;
leakyLMS.w[j] = c0 * leakyLMS.w[j] + c1 * leakyLMS.d[idx];
}
leakyLMS.in_idx = (leakyLMS.in_idx + leakyLMS.mask) & leakyLMS.mask;
}
}
#endif
static void AudioDriver_Dsp_Init(volatile dsp_params_t* dsp_p)
{
dsp_p->active_toggle = 0xff; // used to hold the button G2 "toggle" setting.
// Commented settings below are read from configuration store, no need to initialize them
// dsp_p->active = 0; // TRUE if DSP noise reduction is to be enabled
// dsp_p->nr_strength = 50; // "Strength" of DSP noise reduction (50 = medium)
#ifdef USE_LMS_AUTONOTCH
// dsp_p->notch_numtaps = DSP_NOTCH_NUMTAPS_DEFAULT; // default for number of FFT taps for notch filter
// dsp_p->notch_delaybuf_len = DSP_NOTCH_DELAYBUF_DEFAULT;
// dsp_p->notch_mu = DSP_NOTCH_MU_DEFAULT;
#endif
// dsp_p->notch_frequency = 800; // notch start frequency for manual notch filter
// dsp_p->peak_frequency = 750; // peak start frequency
// dsp_p->nb_setting = 0; // Noise Blanker setting
// dsp_p->bass_gain = 2; // gain of the low shelf EQ filter
// dsp_p->treble_gain = 0; // gain of the high shelf EQ filter
// dsp_p->tx_bass_gain = 4; // gain of the TX low shelf EQ filter
// dsp_p->tx_treble_gain = 4; // gain of the TX high shelf EQ filter
}
/**
* One-time init of FreeDV codec's audio driver part, mostly filters
*/
#define FIR_RX_HILBERT_STATE_SIZE (IQ_RX_NUM_TAPS_MAX + IQ_RX_BLOCK_SIZE)
#ifndef USE_SIMPLE_FREEDV_FILTERS
static float32_t __MCHF_SPECIALMEM Fir_FreeDV_Rx_Hilbert_State_I[FIR_RX_HILBERT_STATE_SIZE];
static float32_t __MCHF_SPECIALMEM Fir_FreeDV_Rx_Hilbert_State_Q[FIR_RX_HILBERT_STATE_SIZE];
static arm_fir_instance_f32 Fir_FreeDV_Rx_Hilbert_I;
static arm_fir_instance_f32 Fir_FreeDV_Rx_Hilbert_Q;
#endif
static void AudioDriver_FreeDV_Rx_Init()
{
#ifdef USE_SIMPLE_FREEDV_FILTERS
IIR_biquad_FreeDV_I.pCoeffs = FreeDV_coeffs[0]; // FreeDV Filter test -DL2FW-
IIR_biquad_FreeDV_Q.pCoeffs = FreeDV_coeffs[0];
#else
#ifdef STM32F4
arm_fir_init_f32(&Fir_FreeDV_Rx_Hilbert_I,IQ_RX_NUM_TAPS_LO,(float32_t*)i_rx_FREEDV_700D_F4_coeffs,Fir_FreeDV_Rx_Hilbert_State_I, IQ_BLOCK_SIZE);
arm_fir_init_f32(&Fir_FreeDV_Rx_Hilbert_Q,IQ_RX_NUM_TAPS_LO,(float32_t*)q_rx_FREEDV_700D_F4_coeffs,Fir_FreeDV_Rx_Hilbert_State_Q, IQ_BLOCK_SIZE);
#else
arm_fir_init_f32(&Fir_FreeDV_Rx_Hilbert_I,IQ_RX_NUM_TAPS,(float32_t*)i_rx_FREEDV_700D_coeffs,Fir_FreeDV_Rx_Hilbert_State_I, IQ_BLOCK_SIZE);
arm_fir_init_f32(&Fir_FreeDV_Rx_Hilbert_Q,IQ_RX_NUM_TAPS,(float32_t*)q_rx_FREEDV_700D_coeffs,Fir_FreeDV_Rx_Hilbert_State_Q, IQ_BLOCK_SIZE);
#endif
#endif
}
void AudioDriver_AgcWdsp_Set()
{
AudioAgc_SetupAgcWdsp(ads.decimated_freq, ts.dmod_mode == DEMOD_AM || ts.dmod_mode == DEMOD_SAM);
}
/**
* Initializes rx audio related data structures, must be called before audio interrupt becomes active
*
* Called once during startup by AudioDriver_Init
*
*/
static void RxProcessor_Init(void)
{
AudioAgc_AgcWdsp_Init(); // RX
NR_Init(); // RX
// also belongs to the NR audio, this is our preprocessing / postprocessing
// never changes, so we place it here
// Set up RX decimation/filter
// this filter instance is also used for Convolution !
arm_fir_decimate_init_f32(&DECIMATE_NR, 4, 2, NR_decimate_coeffs, decimNRState, FIR_RXAUDIO_BLOCK_SIZE);
// should be a very light lowpass @2k7
// this filter instance is also used for Convolution !
arm_fir_interpolate_init_f32(&INTERPOLATE_NR, 2, NR_INTERPOLATE_NO_TAPS, NR_interpolate_coeffs, interplNRState, FIR_RXAUDIO_BLOCK_SIZE);
// should be a very light lowpass @2k7
#ifdef USE_LEAKY_LMS
AudioDriver_LeakyLmsNr_Init(); // RX
#endif
AudioDriver_FreeDV_Rx_Init(); // RX
AudioDriver_FM_Rx_Init(&ads.fm_conf); // RX
// AudioReverb_Init(); //TX
// ads.fade_leveler = 0;
}
/**
* Initializes most of the audio related data structures, must be called before audio interrupt becomes active
* DO NOT ACTIVATE THE INTERRUPT BEFORE AudioDriver_SetProcessingChain() has been called, which handles the dynamic part
* of the initialization based on the mode and filter settings etc.
*
* Do all one time initialization for codecs, (de)modulators, dsp, ... , from here. Nothing here should depend on a specific
* dmod_mode being set. All dmod_mode specific setup/init must be done in AudioDriver_SetProcessingChain() or related functions
*
* Called once during startup
*
*/
void AudioDriver_Init()
{
// Audio filter disabled
// although we are not have the interrupt running, some of the called code may
// refuse to work if dsp.inhibit and/or af_disabled is 0
ts.dsp.inhibit++;
ads.af_disabled++;
// DSP related init
AudioDriver_Dsp_Init(&ts.dsp); // RX/TX
// AUDIO KEY BEEPS (injected into audio output stream)
AudioManagement_KeyBeepPrepare(); // load/set beep frequency
// Codecs/Demod init
Rtty_Modem_Init(ts.samp_rate); // RX/TX
Psk_Modem_Init(ts.samp_rate); // RX/TX
RxProcessor_Init();
TxProcessor_Init();
// Audio filter enabled
ads.af_disabled--;
ts.dsp.inhibit--;
assert(ads.af_disabled == 0);
assert(ts.dsp.inhibit == 0);
}
/**
* Configures SAM PLL according to settings
*/
void AudioDriver_SetSamPllParameters()
{
// definitions and intializations for synchronous AM demodulation = SAM
const float32_t decimSampleRate = ads.decimated_freq;
const float32_t pll_fmax = ads.pll_fmax_int;
// DX adjustments: zeta = 0.15, omegaN = 100.0
// very stable, but does not lock very fast
// standard settings: zeta = 1.0, omegaN = 250.0
// maybe user can choose between slow (DX), medium, fast SAM PLL
// zeta / omegaN
// DX = 0.2, 70
// medium 0.6, 200
// fast 1.0, 500
//ads.zeta_int = 80; // zeta * 100 !!!
// 0.01;// 0.001; // 0.1; //0.65; // PLL step response: smaller, slower response 1.0 - 0.1
//ads.omegaN_int = 250; //200.0; // PLL bandwidth 50.0 - 1000.0
float32_t omegaN = ads.omegaN_int; //200.0; // PLL bandwidth 50.0 - 1000.0
float32_t zeta = (float32_t)ads.zeta_int / 100.0; // 0.01;// 0.001; // 0.1; //0.65; // PLL step response: smaller, slower response 1.0 - 0.1
//pll
adb.sam.omega_min = - (2.0 * PI * pll_fmax / decimSampleRate);
adb.sam.omega_max = (2.0 * PI * pll_fmax / decimSampleRate);
adb.sam.g1 = (1.0 - expf(-2.0 * omegaN * zeta / decimSampleRate));
adb.sam.g2 = (- adb.sam.g1 + 2.0 * (1 - expf(- omegaN * zeta / decimSampleRate)
* cosf(omegaN / decimSampleRate * sqrtf(1.0 - zeta * zeta)))); //0.01036367597097734813032783691644; //
//fade leveler
float32_t tauR = 0.02; // value emperically determined
float32_t tauI = 1.4; // value emperically determined
adb.sam.mtauR = (expf(- 1 / (decimSampleRate * tauR)));
adb.sam.onem_mtauR = (1.0 - adb.sam.mtauR);
adb.sam.mtauI = (expf(- 1 / (decimSampleRate * tauI)));
adb.sam.onem_mtauI = (1.0 - adb.sam.mtauI);
}
static void AudioDriver_SetRxIqCorrection(void)
{
// these change during operation
adb.iq_corr.M_c1 = 0.0;
adb.iq_corr.M_c2 = 1.0;
adb.iq_corr.teta1_old = 0.0;
adb.iq_corr.teta2_old = 0.0;
adb.iq_corr.teta3_old = 0.0;
}
/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
* Cascaded biquad (notch, peak, lowShelf, highShelf) [DD4WH, april 2016]
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/
// biquad_1 : Notch & peak filters & lowShelf (Bass) in the decimated path
// biquad 2 : Treble in the 48kHz path
// TX_biquad: Bass & Treble in the 48kHz path
// DSP Audio-EQ-cookbook for generating the coeffs of the filters on the fly
// www.musicdsp.org/files/Audio-EQ-Cookbook.txt [by Robert Bristow-Johnson]
//
// the ARM algorithm assumes the biquad form
// y[n] = coeffs[B0] * x[n] + coeffs[B1] * x[n-1] + coeffs[B2] * x[n-2] + coeffs[A1] * y[n-1] + a2 * y[n-2]
//
// However, the cookbook formulae by Robert Bristow-Johnson AND the Iowa Hills IIR Filter designer
// use this formula:
//
// y[n] = coeffs[B0] * x[n] + coeffs[B1] * x[n-1] + coeffs[B2] * x[n-2] - coeffs[A1] * y[n-1] - coeffs[A2] * y[n-2]
//
// Therefore, we have to use negated coeffs[A1] and coeffs[A2] for use with the ARM function
// notch implementation
//
// we also have to divide every coefficient by a0 !
// y[n] = coeffs[B0]/a0 * x[n] + coeffs[B1]/a0 * x[n-1] + coeffs[B2]/a0 * x[n-2] - coeffs[A1]/a0 * y[n-1] - coeffs[A2]/a0 * y[n-2]
//
//
#define B0 0
#define B1 1
#define B2 2
#define A1 3
#define A2 4
/**
* @brief Biquad Filter Init Helper function which copies the biquad coefficients into the filter array itself
*/
void AudioDriver_SetBiquadCoeffs(float32_t* coeffsTo,const float32_t* coeffsFrom)
{
coeffsTo[0] = coeffsFrom[0];
coeffsTo[1] = coeffsFrom[1];
coeffsTo[2] = coeffsFrom[2];
coeffsTo[3] = coeffsFrom[3];
coeffsTo[4] = coeffsFrom[4];
}
/**
* @brief Biquad Filter Init Helper function which copies the biquad coefficients into all filter instances of the 1 or 2 channel audio
* @param biquad_inst_array a pointer to an either 1 or 2 element sized array of biquad filter instances. Make sure the array has the expected size!
* @param idx of first element of stage coefficients (i.e. 0, 5, 10, ... ) since we have 5 element coeff arrays per stage
*/
void AudioDriver_SetBiquadCoeffsAllInstances(arm_biquad_casd_df1_inst_f32 biquad_inst_array[NUM_AUDIO_CHANNELS], uint32_t idx, const float32_t* coeffsFrom)
{
for (int chan = 0; chan < NUM_AUDIO_CHANNELS; chan++)
{
AudioDriver_SetBiquadCoeffs(&biquad_inst_array[chan].pCoeffs[idx],coeffsFrom);
}
}
/**
* @brief Biquad Filter Init Helper function which applies the filter specific scaling to calculated coefficients
*/
void AudioDriver_ScaleBiquadCoeffs(float32_t coeffs[5],const float32_t scalingA, const float32_t scalingB)
{
coeffs[A1] = coeffs[A1] / scalingA;
coeffs[A2] = coeffs[A2] / scalingA;
coeffs[B0] = coeffs[B0] / scalingB;
coeffs[B1] = coeffs[B1] / scalingB;
coeffs[B2] = coeffs[B2] / scalingB;
}
/**
* @brief Biquad Filter Init Helper function to calculate a notch filter aka narrow bandstop filter
*/
void AudioDriver_CalcBandstop(float32_t coeffs[5], float32_t f0, float32_t FS)
{
float32_t Q = 10; // larger Q gives narrower notch
float32_t w0 = 2 * PI * f0 / FS;
float32_t alpha = sinf(w0) / (2 * Q);
coeffs[B0] = 1;
coeffs[B1] = - 2 * cosf(w0);
coeffs[B2] = 1;
float32_t scaling = 1 + alpha;
coeffs[A1] = 2 * cosf(w0); // already negated!
coeffs[A2] = alpha - 1; // already negated!
AudioDriver_ScaleBiquadCoeffs(coeffs,scaling, scaling);
}
/**
* @brief Biquad Filter Init Helper function to calculate a peak filter aka a narrow bandpass filter
*/
void AudioDriver_CalcBandpass(float32_t coeffs[5], float32_t f0, float32_t FS)
{
/* // peak filter = peaking EQ
f0 = ts.dsp.peak_frequency;
//Q = 15; //
// bandwidth in octaves between midpoint (Gain / 2) gain frequencies
float32_t BW = 0.05;
w0 = 2 * PI * f0 / FSdec;
//alpha = sin(w0) / (2 * Q);
alpha = sin (w0) * sinh( log(2) / 2 * BW * w0 / sin(w0) );
float32_t Gain = 12;
A = powf(10.0, (Gain/40.0));
coeffs[B0] = 1 + (alpha * A);
coeffs[B1] = - 2 * cos(w0);
coeffs[B2] = 1 - (alpha * A);
a0 = 1 + (alpha / A);
coeffs[A1] = 2 * cos(w0); // already negated!
coeffs[A2] = (alpha/A) - 1; // already negated!
*/
/* // test the BPF coefficients, because actually we want a "peak" filter without gain!
// Bandpass filter constant 0dB peak gain
// this filter was tested: "should have more gain and less Q"
f0 = ts.dsp.peak_frequency;
Q = 20; //
w0 = 2 * PI * f0 / FSdec;
alpha = sin(w0) / (2 * Q);
// A = 1; // gain = 1
// A = 3; // 10^(10/40); 15dB gain
coeffs[B0] = alpha;
coeffs[B1] = 0;
coeffs[B2] = - alpha;
a0 = 1 + alpha;
coeffs[A1] = 2 * cos(w0); // already negated!
coeffs[A2] = alpha - 1; // already negated!
*/
// BPF: constant skirt gain, peak gain = Q
float32_t Q = 4; //
float32_t BW = 0.03;
float32_t w0 = 2 * PI * f0 / FS;
float32_t alpha = sinf (w0) * sinhf( log(2) / 2 * BW * w0 / sinf(w0) ); //
coeffs[B0] = Q * alpha;
coeffs[B1] = 0;
coeffs[B2] = - Q * alpha;
float32_t scaling = 1 + alpha;
coeffs[A1] = 2 * cosf(w0); // already negated!
coeffs[A2] = alpha - 1; // already negated!
AudioDriver_ScaleBiquadCoeffs(coeffs,scaling, scaling);
}
/**
* @brief Biquad Filter Init Helper function to calculate a treble adjustment filter aka high shelf filter
*/
void AudioDriver_CalcHighShelf(float32_t coeffs[5], float32_t f0, float32_t S, float32_t gain, float32_t FS)
{
float32_t w0 = 2 * PI * f0 / FS;
float32_t A = pow10f(gain/40.0); // gain ranges from -20 to 5
float32_t alpha = sinf(w0) / 2 * sqrtf( (A + 1/A) * (1/S - 1) + 2 );
float32_t cosw0 = cosf(w0);
float32_t twoAa = 2 * sqrtf(A) * alpha;
// highShelf
//
coeffs[B0] = A * ( (A + 1) + (A - 1) * cosw0 + twoAa );
coeffs[B1] = - 2 * A * ( (A - 1) + (A + 1) * cosw0 );
coeffs[B2] = A * ( (A + 1) + (A - 1) * cosw0 - twoAa );
float32_t scaling = (A + 1) - (A - 1) * cosw0 + twoAa ;
coeffs[A1] = - 2 * ( (A - 1) - (A + 1) * cosw0 ); // already negated!
coeffs[A2] = twoAa - (A + 1) + (A - 1) * cosw0; // already negated!
// DCgain = 2; //
// DCgain = (coeffs[B0] + coeffs[B1] + coeffs[B2]) / (1 - (- coeffs[A1] - coeffs[A2])); // takes into account that coeffs[A1] and coeffs[A2] are already negated!
float32_t DCgain = 1.0 * scaling;
AudioDriver_ScaleBiquadCoeffs(coeffs,scaling, DCgain);
}
void AudioDriver_CalcPeakEQ(float32_t coeffs[5], float32_t f0, float32_t q, float32_t gain, float32_t FS)
{
float32_t w0 = 2 * PI * f0 / FS;
float32_t A = pow10f(gain/40.0);
float32_t alpha = sinf(w0) / (2 * q);
float32_t cosw0 = cosf(w0);
coeffs[B0] = 1 + alpha * A;
coeffs[B1] = -2 * cosw0;
coeffs[B2] = 1 - alpha * A;
float32_t scaling = 1 + alpha / A;
coeffs[A1] = 2 * cosw0;
coeffs[A2] = alpha/A - 1;
float32_t DCgain = 1.0 * scaling;
AudioDriver_ScaleBiquadCoeffs(coeffs,scaling, DCgain);
}
/**
* @brief Biquad Filter Init Helper function to calculate a bass adjustment filter aka low shelf filter
*/
void AudioDriver_CalcLowShelf(float32_t coeffs[5], float32_t f0, float32_t S, float32_t gain, float32_t FS)
{
float32_t w0 = 2 * PI * f0 / FS;
float32_t A = pow10f(gain/40.0); // gain ranges from -20 to 5
float32_t alpha = sinf(w0) / 2 * sqrtf( (A + 1/A) * (1/S - 1) + 2 );
float32_t cosw0 = cosf(w0);
float32_t twoAa = 2 * sqrtf(A) * alpha;
// lowShelf
coeffs[B0] = A * ( (A + 1) - (A - 1) * cosw0 + twoAa );
coeffs[B1] = 2 * A * ( (A - 1) - (A + 1) * cosw0 );
coeffs[B2] = A * ( (A + 1) - (A - 1) * cosw0 - twoAa );
float32_t scaling = (A + 1) + (A - 1) * cosw0 + twoAa ;
coeffs[A1] = 2 * ( (A - 1) + (A + 1) * cosw0 ); // already negated!
coeffs[A2] = twoAa - (A + 1) - (A - 1) * cosw0; // already negated!
// scaling the feedforward coefficients for gain adjustment !
// "DC gain of an IIR filter is the sum of the filters<72> feedforward coeffs divided by
// 1 minus the sum of the filters<72> feedback coeffs" (Lyons 2011)
// float32_t DCgain = (coeffs[B0] + coeffs[B1] + coeffs[B2]) / (1 - (coeffs[A1] + coeffs[A2]));
// does not work for some reason?
// I take a divide by a constant instead !
// DCgain = (coeffs[B0] + coeffs[B1] + coeffs[B2]) / (1 - (- coeffs[A1] - coeffs[A2])); // takes into account that coeffs[A1] and coeffs[A2] are already negated!
float32_t DCgain = 1.0 * scaling; //
AudioDriver_ScaleBiquadCoeffs(coeffs,scaling, DCgain);
}
#if 0
/**
* @brief Biquad Filter Init Helper function to calculate a notch filter aka narrow bandstop filter with variable bandwidth
*/
static void AudioDriver_CalcNotch(float32_t coeffs[5], float32_t f0, float32_t BW, float32_t FS)
{
float32_t w0 = 2 * PI * f0 / FS;
float32_t alpha = sinf(w0)*sinh( logf(2.0)/2.0 * BW * w0/sinf(w0) );
coeffs[B0] = 1;
coeffs[B1] = - 2 * cosf(w0);
coeffs[B2] = 1;
float32_t scaling = 1 + alpha;
coeffs[A1] = 2 * cosf(w0); // already negated!
coeffs[A2] = alpha - 1; // already negated!
AudioDriver_ScaleBiquadCoeffs(coeffs,scaling, scaling);
}
#endif
static const float32_t biquad_passthrough[] = { 1, 0, 0, 0, 0 };
/**
* @brief Filter Init used for processing audio for RX and TX
*/
static void AudioDriver_SetRxTxAudioProcessingAudioFilters(uint8_t dmod_mode)
{
float32_t FSdec = AUDIO_SAMPLE_RATE / (ts.filters_p->sample_rate_dec != 0 ? ts.filters_p->sample_rate_dec : 1) ;
// the notch filter is in biquad 1 and works at the decimated sample rate FSdec
float32_t coeffs[5];
const float32_t *coeffs_ptr;
// setting the Coefficients in the notch filter instance
if (is_dsp_mnotch())
{
AudioDriver_CalcBandstop(coeffs, ts.dsp.notch_frequency , FSdec);
coeffs_ptr = coeffs;
}
else // passthru
{
coeffs_ptr = biquad_passthrough;
}
AudioDriver_SetBiquadCoeffsAllInstances(IIR_biquad_1, 0, coeffs_ptr);
// this is an auto-notch-filter detected by the NR algorithm
// biquad 1, 4th stage
// if(0) // temporarily deactivated
// if((is_dsp_notch()) && (ts.filters_p->sample_rate_dec == RX_DECIMATION_RATE_12KHZ))
AudioDriver_SetBiquadCoeffsAllInstances(IIR_biquad_1, 15, biquad_passthrough);
// the peak filter is in biquad 1 and works at the decimated sample rate FSdec
if(is_dsp_mpeak())
{
AudioDriver_CalcBandpass(coeffs, ts.dsp.peak_frequency, FSdec);
coeffs_ptr = coeffs;
}
else //passthru
{
coeffs_ptr = biquad_passthrough;
}
AudioDriver_SetBiquadCoeffsAllInstances(IIR_biquad_1, 5, coeffs_ptr);
// EQ shelving filters
//
// the bass filter is in biquad 1 and works at the decimated sample rate FSdec
//
// Bass / lowShelf
AudioDriver_CalcLowShelf(coeffs, 250, 0.7, ts.dsp.bass_gain, FSdec);
AudioDriver_SetBiquadCoeffsAllInstances(IIR_biquad_1, 10, coeffs);
// Treble = highShelf
// the treble filter is in biquad 2 and works at 48000ksps
AudioDriver_CalcHighShelf(coeffs, 3500, 0.9, ts.dsp.treble_gain, AUDIO_SAMPLE_RATE);
AudioDriver_SetBiquadCoeffsAllInstances(IIR_biquad_2, 0, coeffs);
TxProcessor_Set(dmod_mode);
}
/**
* Sets up the spectrum filters according to the current zoom level
*/
static void AudioDriver_Spectrum_Set(void)
{
// this sets the coefficients for the ZoomFFT decimation filter
// according to the desired magnification mode sd.magnify.
// The magnification is 2^sd.magnify (0 - 5 -> 1x - 32x)
if(sd.magnify > MAGNIFY_MAX)
{
sd.magnify = MAGNIFY_MIN;
}
// for 0 the mag_coeffs will a NULL ptr, since the filter is not going to be used in this mode!
IIR_biquad_Zoom_FFT_I.pCoeffs = mag_coeffs[sd.magnify];
IIR_biquad_Zoom_FFT_Q.pCoeffs = mag_coeffs[sd.magnify];
// Set up ZOOM FFT FIR decimation filters
// switch right FIR decimation filter depending on sd.magnify
arm_fir_decimate_init_f32(&DECIMATE_ZOOM_FFT_I,
FirZoomFFTDecimate[sd.magnify].numTaps,
(1 << sd.magnify), // Decimation factor
FirZoomFFTDecimate[sd.magnify].pCoeffs,
decimZoomFFTIState, // Filter state variables
FIR_RXAUDIO_BLOCK_SIZE);
arm_fir_decimate_init_f32(&DECIMATE_ZOOM_FFT_Q,
FirZoomFFTDecimate[sd.magnify].numTaps,
(1 << sd.magnify), // Decimation factor
FirZoomFFTDecimate[sd.magnify].pCoeffs,
decimZoomFFTQState, // Filter state variables
FIR_RXAUDIO_BLOCK_SIZE);
}
/**
* @brief configures filters/dsp etc. so that audio processing works according to the current configuration
* @param dmod_mode needs to know the demodulation mode
* @param reset_dsp_nr whether it is supposed to reset also DSP related filters (in most cases false is to be used here)
*/
void AudioDriver_SetProcessingChain(uint8_t dmod_mode, bool reset_dsp_nr)
{
ts.dsp.inhibit++;
ads.af_disabled++;
// make sure we have a proper filter path for the given mode
// the commented out part made the code only look at last used/selected filter path if the current filter path is not applicable
// with it commented out the filter path is ALWAYS loaded from the last used/selected memory
// I.e. setting the ts.filter_path anywere else in the code is useless. You have to call AudioFilter_NextApplicableFilterPath in order to
// select a new filter path as this sets the last used/selected memory for a demod mode.
ts.filter_path = AudioFilter_NextApplicableFilterPath(PATH_ALL_APPLICABLE|PATH_LAST_USED_IN_MODE,AudioFilter_GetFilterModeFromDemodMode(dmod_mode),ts.filter_path);
const FilterPathDescriptor* filters_p = ts.filters_p = &FilterPathInfo[ts.filter_path];
// Adjust decimation rate based on selected filter
ads.decimation_rate = filters_p->sample_rate_dec;
ads.decimated_freq = IQ_SAMPLE_RATE / ads.decimation_rate;
for (int chan = 0; chan < NUM_AUDIO_CHANNELS; chan++)
{
if (ts.filters_p->pre_instance != NULL)
{
// if we turn on a filter, set the number of members to the number of elements last
IIR_PreFilter[chan].pkCoeffs = filters_p->pre_instance->pkCoeffs; // point to reflection coefficients
IIR_PreFilter[chan].pvCoeffs = filters_p->pre_instance->pvCoeffs; // point to ladder coefficients
IIR_PreFilter[chan].numStages = filters_p->pre_instance->numStages; // number of stages
}
else
{
// if we turn off a filter, set the number of members to 0 first
IIR_PreFilter[chan].numStages = 0; // number of stages
IIR_PreFilter[chan].pkCoeffs = NULL; // point to reflection coefficients
IIR_PreFilter[chan].pvCoeffs = NULL; // point to ladder coefficients
}
// Initialize IIR filter state buffer
arm_fill_f32(0.0,iir_rx_state[chan],IIR_RX_STATE_ARRAY_SIZE);
IIR_PreFilter[chan].pState = iir_rx_state[chan]; // point to state array for IIR filter
// Initialize IIR antialias filter state buffer
if (filters_p->iir_instance != NULL)
{
// if we turn on a filter, set the number of members to the number of elements last
IIR_AntiAlias[chan].pkCoeffs = filters_p->iir_instance->pkCoeffs; // point to reflection coefficients
IIR_AntiAlias[chan].pvCoeffs = filters_p->iir_instance->pvCoeffs; // point to ladder coefficients
IIR_AntiAlias[chan].numStages = filters_p->iir_instance->numStages; // number of stages
}
else
{
// if we turn off a filter, set the number of members to 0 first
IIR_AntiAlias[chan].numStages = 0;
IIR_AntiAlias[chan].pkCoeffs = NULL;
IIR_AntiAlias[chan].pvCoeffs = NULL;
}
arm_fill_f32(0.0,iir_aa_state[chan],IIR_RX_STATE_ARRAY_SIZE);
IIR_AntiAlias[chan].pState = iir_aa_state[chan]; // point to state array for IIR filter
}
// TODO: We only have to do this, if the audio signal filter configuration changes
// RX+ TX Bass, Treble, Peak, Notch
AudioDriver_SetRxTxAudioProcessingAudioFilters(dmod_mode);
// initialize the goertzel filter used to detect CW signals at a given frequency in the audio stream
CwDecode_Filter_Set();
/*+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
* End of coefficient calculation and setting for cascaded biquad
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++*/
#ifdef USE_LMS_AUTONOTCH
// AUTO NOTCH INIT START
// LMS instance 2 - Automatic Notch Filter
// Calculate "mu" (convergence rate) from user "Notch ConvRate" setting
float32_t mu_calc = log10f(((ts.dsp.notch_mu + 1.0)/1500.0) + 1.0); // get user setting (0 = slowest)
// use "canned" init to initialize the filter coefficients
arm_lms_norm_init_f32(&lmsData.lms2Norm_instance, ts.dsp.notch_numtaps, lmsData.lms2NormCoeff_f32, lmsData.lms2StateF32, mu_calc, IQ_BLOCK_SIZE);
arm_fill_f32(0.0,lmsData.lms2_nr_delay,DSP_NOTCH_BUFLEN_MAX);
if(reset_dsp_nr) // are we to reset the coefficient buffer as well?
{
arm_fill_f32(0.0,lmsData.lms2NormCoeff_f32,DSP_NOTCH_NUMTAPS_MAX); // yes - zero coefficient buffers
}
if((ts.dsp.notch_delaybuf_len > DSP_NOTCH_BUFLEN_MAX) || (ts.dsp.notch_delaybuf_len < DSP_NOTCH_BUFLEN_MIN))
{
ts.dsp.notch_delaybuf_len = DSP_NOTCH_DELAYBUF_DEFAULT;
}
// AUTO NOTCH INIT END
#endif
// NEW SPECTRAL NOISE REDUCTION
// convert user setting of noise reduction to alpha NR parameter
// alpha ranges from 0.9 to 0.999 [float32_t]
// dsp_nr_strength is from 0 to 100 [uint8_t]
// nr_params.alpha = 0.899 + ((float32_t)ts.dsp.nr_strength / 1000.0);
nr_params.alpha = 0.799 + ((float32_t)ts.dsp.nr_strength / 1000.0);
// NEW AUTONOTCH
// not yet working !
// set to passthrough
//AudioNr_ActivateAutoNotch(0, 0);
AudioDriver_Spectrum_Set();
#ifdef USE_CONVOLUTION
// TODO: insert interpolation filter settings for convolution filter HERE
#else
// Set up RX interpolation/filter
// NOTE: Phase Length MUST be an INTEGER and is the number of taps divided by the decimation rate, and it must be greater than 1.
for (int chan = 0; chan < NUM_AUDIO_CHANNELS; chan++)
{
if (filters_p->interpolate != NULL)
{
arm_fir_interpolate_init_f32(&INTERPOLATE_RX[chan],
ads.decimation_rate,
filters_p->interpolate->phaseLength,
filters_p->interpolate->pCoeffs,
interpState[chan], IQ_BLOCK_SIZE);
}
else
{
INTERPOLATE_RX[chan].phaseLength = 0;
INTERPOLATE_RX[chan].pCoeffs = NULL;
}
}
#endif
#ifdef USE_CONVOLUTION
// Convolution Filter
// calculate coeffs
// for first trial, use hard-coded USB filter from 250Hz to 2700Hz
// hardcoded sample rate and Blackman-Harris 4th term
cbs.size = 128; // 128 samples processed at one time, this defines the latency:
// latency = cbs.size / sample rate = 128/48000 = 2.7ms
cbs.nc = 1024; // use 1024 coefficients
cbs.nfor = cbs.nc / cbs.size; // number of blocks used for the uniformly partitioned convolution
cbs.buffidx = 0; // needs to be reset to zero each time new coeffs are being calculated
AudioDriver_CalcConvolutionFilterCoeffs (cbs.nc, 250.0, 2700.0, IQ_SAMPLE_RATE_F, 0, 1, 1.0);
#endif
AudioDriver_SetSamPllParameters();
AudioDriver_SetRxIqCorrection();
AudioFilter_SetRxHilbertAndDecimationFIR(dmod_mode); // this switches the Hilbert/FIR-filters
AudioDriver_AgcWdsp_Set();
// Unlock - re-enable filtering
if (ads.af_disabled) { ads.af_disabled--; }
if (ts.dsp.inhibit) { ts.dsp.inhibit--; }
}
#if USE_OBSOLETE_NOISEBLANKER
//*----------------------------------------------------------------------------
//* Function Name : audio_rx_noise_blanker [KA7OEI]
//* Object : noise blanker
//* Object :
//* Input Parameters : I/Q 16 bit audio data, size of buffer
//* Output Parameters :
//* Functions called :
//*----------------------------------------------------------------------------
#define AUDIO_RX_NB_DELAY_BUFFER_ITEMS (16)
#define AUDIO_RX_NB_DELAY_BUFFER_SIZE (AUDIO_RX_NB_DELAY_BUFFER_ITEMS*2)
static void AudioDriver_NoiseBlanker(AudioSample_t * const src, int16_t blockSize)
{
static int16_t delay_buf[AUDIO_RX_NB_DELAY_BUFFER_SIZE];
static uchar delbuf_inptr = 0, delbuf_outptr = 2;
ulong i;
float sig;
float nb_short_setting;
// static float avg_sig;
static uchar nb_delay = 0;
static float nb_agc = 0;
if((ts.nb_setting > 0) && (ts.dsp.active & DSP_NB_ENABLE)
// && (ts.dmod_mode != DEMOD_AM && ts.dmod_mode != DEMOD_FM)
&& (ts.dmod_mode != DEMOD_FM))
// && (filters_p->sample_rate_dec != RX_DECIMATION_RATE_24KHZ ))
// bail out if noise blanker disabled, in AM/FM mode, or set to 10 kHz
{
nb_short_setting = ts.nb_setting; // convert and rescale NB1 setting for higher resolution in adjustment
nb_short_setting /= 2;
for(i = 0; i < blockSize; i++) // Noise blanker function
{
sig = fabs(src[i].l); // get signal amplitude. We need only look at one of the two audio channels since they will be the same.
sig /= ads.codec_gain_calc; // Scale for codec A/D gain adjustment
// avg_sig = (avg_sig * NB_AVG_WEIGHT) + ((float)(*src) * NB_SIG_WEIGHT); // IIR-filtered short-term average signal level (e.g. low-pass audio)
delay_buf[delbuf_inptr++] = src[i].l; // copy first byte into delay buffer
delay_buf[delbuf_inptr++] = src[i].r; // copy second byte into delay buffer
nb_agc = (ads.nb_agc_filt * nb_agc) + (ads.nb_sig_filt * sig); // IIR-filtered "AGC" of current overall signal level
if(((sig) > (nb_agc * (((MAX_NB_SETTING/2) + 1.75) - nb_short_setting))) && (nb_delay == 0)) // did a pulse exceed the threshold?
{
nb_delay = AUDIO_RX_NB_DELAY_BUFFER_ITEMS; // yes - set the blanking duration counter
}
if(!nb_delay) // blank counter not active
{
src[i].l = delay_buf[delbuf_outptr++]; // pass through delayed audio, unchanged
src[i].r = delay_buf[delbuf_outptr++];
}
else // It is within the blanking pulse period
{
src[i].l = 0; // (int16_t)avg_sig; // set the audio buffer to "mute" during the blanking period
src[i].r = 0; //(int16_t)avg_sig;
nb_delay--; // count down the number of samples that we are to blank
}
// RINGBUFFER
delbuf_outptr %= AUDIO_RX_NB_DELAY_BUFFER_SIZE;
delbuf_inptr %= AUDIO_RX_NB_DELAY_BUFFER_SIZE;
}
}
}
#endif
#ifdef USE_FREEDV
/**
* @returns: true if digital signal should be used (no analog processing should be done), false -> analog processing maybe used
* since no digital signal was detected.
*/
static bool AudioDriver_RxProcessorFreeDV (iq_buffer_t* iq_buf_p, float32_t * const dst, int16_t blockSize)
{
bool retval = false;
// Freedv DL2FW
static int16_t modulus_Decimate = 0, modulus_Interpolate = 0;
static bool bufferFilled;
static float32_t History[3]={0.0,0.0,0.0};
const int32_t factor_Decimate = IQ_SAMPLE_RATE/8000; // 6x @ 48ksps
const int32_t factor_Interpolate = AUDIO_SAMPLE_RATE/8000; // 6x @ 48ksps
bool lsb_active = RadioManagement_LSBActive(ts.dmod_mode);
// we switch I and Q here so that we later always use USB demodulation
// this works because we have the frequency of interest centered.
float32_t* real = (lsb_active == true)? iq_buf_p->q_buffer : iq_buf_p->i_buffer;
float32_t* imag = (lsb_active == true)? iq_buf_p->i_buffer : iq_buf_p->q_buffer;
float32_t real_buffer[blockSize];
float32_t imag_buffer[blockSize];
#ifdef USE_SIMPLE_FREEDV_FILTERS
arm_biquad_cascade_df1_f32 (&IIR_biquad_FreeDV_I, real, real_buffer, blockSize);
arm_biquad_cascade_df1_f32 (&IIR_biquad_FreeDV_Q, imag, imag_buffer, blockSize);
#else
// we run a hilbert transform including a low pass to avoid
// aliasing artifacts
arm_fir_f32(&Fir_FreeDV_Rx_Hilbert_I,real,real_buffer,blockSize);
arm_fir_f32(&Fir_FreeDV_Rx_Hilbert_Q,imag,imag_buffer,blockSize);
#endif
// DOWNSAMPLING
for (int k = 0; k < blockSize; k++)
{
if (modulus_Decimate == 0) //every nth sample has to be catched -> downsampling
{
const float32_t f32_to_i16_gain = 10;
// 10 for normal RX, 1000 for USB PC debugging
#ifdef USE_SIMPLE_FREEDV_FILTERS
fdv_iq_rb_item_t sample;
// this is the USB demodulation I + Q
sample.real = real_buffer[k] * f32_to_i16_gain;
sample.imag = imag_buffer[k] * f32_to_i16_gain;
RingBuffer_PutSamples(&fdv_iq_rb, &sample, 1);
#else
int16_t sample;
// this is the USB demodulation I + Q
sample = (real_buffer[k] + imag_buffer[k]) * f32_to_i16_gain;
RingBuffer_PutSamples(&fdv_demod_rb, &sample, 1);
#endif
}
// increment and wrap
modulus_Decimate++;
if (modulus_Decimate == factor_Decimate)
{
modulus_Decimate = 0;
}
}
// if we run out of buffers lately
// we wait for availability of at least 2 buffers
// so that in theory we have uninterrupted flow of audio
// albeit with a delay of 80ms
// we wait for at least one iq frame plus one speech block being processed until we start
// sending audio to the speaker
if (bufferFilled == false && RingBuffer_GetData(&fdv_audio_rb) > FreeDV_Iq_Get_FrameLen())
{
bufferFilled = true;
}
if (bufferFilled == true && RingBuffer_GetData(&fdv_audio_rb) >= (modulus_Interpolate != 0?5:6)) // freeDV encode has finished (running in ui_driver.c)?
{
// Best thing here would be to use the arm_fir_decimate function! Why?
// --> we need phase linear filters, because we have to filter I & Q and preserve their phase relationship
// IIR filters are power saving, but they do not care about phase, so useless at this point
// FIR filters are phase linear, but need processor power
// so we now use the decimation function that upsamples like the code below, BUT at the same time filters
// (and the routine knows that it does not have to multiply with 0 while filtering: if we do upsampling and subsequent
// filtering, the filter does not know that and multiplies with zero 5 out of six times --> very inefficient)
// BUT: we cannot use the ARM function, because decimation factor (6) has to be an integer divide of
// block size (which is 32 in our case --> 32 / 6 = non-integer!)
for (int j=0; j < blockSize; j++) //upsampling with integrated interpolation-filter for M=6
// avoiding multiplications by zero within the arm_iir_filter
{
static float32_t sample;
if (modulus_Interpolate == 0)
{
int16_t sample_in;
RingBuffer_GetSamples(&fdv_audio_rb, &sample_in, 1);
sample = (float32_t)sample_in * 0.25;
// we scale samples down to align this with other demodulators
// TODO: find out correct scaling val, not just a rough estimation
}
dst[j] = Fir_Rx_FreeDV_Interpolate_Coeffs[5-modulus_Interpolate] * History[0] +
Fir_Rx_FreeDV_Interpolate_Coeffs[11-modulus_Interpolate] * History[1] +
Fir_Rx_FreeDV_Interpolate_Coeffs[17-modulus_Interpolate] * History[2] +
Fir_Rx_FreeDV_Interpolate_Coeffs[23-modulus_Interpolate] * sample;
History[0] = History[1];
History[1] = History[2];
History[2] = sample;
// increment and wrap
modulus_Interpolate++;
if (modulus_Interpolate == factor_Interpolate)
{
modulus_Interpolate = 0;
}
}
retval = true; // yes we have output to share
}
else
{
bufferFilled = false;
profileEvent(FreeDVTXUnderrun);
}
if (retval == false && freedv_conf.mute_if_squelched == true)
{
memset(dst,0,sizeof(dst[0])*blockSize);
retval = true;
}
return retval;
}
#endif
// RTTY Experiment based on code from the DSP Tutorial at http://dp.nonoo.hu/projects/ham-dsp-tutorial/18-rtty-decoder-using-iir-filters/
// Used with permission from Norbert Varga, HA2NON under GPLv3 license
#ifdef USE_RTTY_PROCESSOR
static void AudioDriver_RxProcessor_Rtty(float32_t * const src, int16_t blockSize)
{
for (uint16_t idx = 0; idx < blockSize; idx++)
{
Rtty_Demodulator_ProcessSample(src[idx]);
}
}
#endif
static void AudioDriver_RxProcessor_Bpsk(float32_t * const src, int16_t blockSize)
{
for (uint16_t idx = 0; idx < blockSize; idx++)
{
Psk_Demodulator_ProcessSample(src[idx]);
}
}
// FM Demodulator parameters
#define FM_DEMOD_COEFF1 PI/4 // Factors used in arctan approximation used in FM demodulator
#define FM_DEMOD_COEFF2 PI*0.75
//
#define FM_RX_SCALING_2K5 10000 // 33800 // Amplitude scaling factor of demodulated FM audio (normalized for +/- 2.5 kHz deviation at 1 kHZ)
#define FM_RX_SCALING_5K (FM_RX_SCALING_2K5/2) // Amplitude scaling factor of demodulated FM audio (normalized for +/- 5 kHz deviation at 1 kHz)
//
#define FM_AGC_SCALING 2 // Scaling factor for AGC result when in FM (AGC used only for S-meter)
//
#define FM_RX_LPF_ALPHA 0.05 // For FM demodulator: "Alpha" (low-pass) factor to result in -6dB "knee" at approx. 270 Hz
//
#define FM_RX_HPF_ALPHA 0.96 // For FM demodulator: "Alpha" (high-pass) factor to result in -6dB "knee" at approx. 180 Hz
//
#define FM_RX_SQL_SMOOTHING 0.005 // Smoothing factor for IIR squelch noise averaging
#define FM_SQUELCH_HYSTERESIS 3 // Hysteresis for FM squelch
#define FM_SQUELCH_PROC_DECIMATION ((uint32_t)(1/FM_RX_SQL_SMOOTHING)) // Number of times we go through the FM demod algorithm before we do a squelch calculation
#define FM_TONE_DETECT_ALPHA 0.9 // setting for IIR filtering of ratiometric result from frequency-differential tone detection
#define FM_SUBAUDIBLE_TONE_DET_THRESHOLD 1.75 // threshold of "smoothed" output of Goertzel, above which a tone is considered to be "provisionally" detected pending debounce
#define FM_SUBAUDIBLE_DEBOUNCE_MAX 5 // maximum "detect" count in debounce
#define FM_SUBAUDIBLE_TONE_DEBOUNCE_THRESHOLD 2 // number of debounce counts at/above which a tone detection is considered valid
typedef struct
{
float i_prev;
float32_t q_prev;
float32_t lpf_prev;
float32_t hpf_prev_a;
float32_t hpf_prev_b;// used in FM detection and low/high pass processing
float subdet; // used for tone detection
uint8_t count;
uint8_t tdet;// used for squelch processing and debouncing tone detection, respectively
ulong gcount; // used for averaging in tone detection
} demod_fm_data_t;
demod_fm_data_t fm_data;
/**
* FM Demodulator, runs at IQ_SAMPLE_RATE and outputs at same rate, does subtone detection
*
* @author KA7OEI
*
* @param i_buffer
* @param q_buffer
* @param a_buffer
* @param blockSize
* @return
*/
static bool AudioDriver_DemodFM(const float32_t* i_buffer, const float32_t* q_buffer, float32_t* a_buffer, const int16_t blockSize)
{
float32_t goertzel_buf[blockSize], squelch_buf[blockSize];
if (ts.iq_freq_mode != FREQ_IQ_CONV_MODE_OFF)// bail out if translate mode is not active
{
bool tone_det_enabled = ads.fm_conf.subaudible_tone_det_freq != 0;// set a quick flag for checking to see if tone detection is enabled
for (uint16_t i = 0; i < blockSize; i++)
{
// first, calculate "x" and "y" for the arctan2, comparing the vectors of present data with previous data
float32_t y = (fm_data.i_prev * q_buffer[i]) - (i_buffer[i] * fm_data.q_prev);
float32_t x = (fm_data.i_prev * i_buffer[i]) + (q_buffer[i] * fm_data.q_prev);
float32_t angle = atan2f(y, x);
// we now have our audio in "angle"
squelch_buf[i] = angle; // save audio in "d" buffer for squelch noise filtering/detection - done later
// Now do integrating low-pass filter to do FM de-emphasis
float32_t a = fm_data.lpf_prev + (FM_RX_LPF_ALPHA * (angle - fm_data.lpf_prev)); //
fm_data.lpf_prev = a; // save "[n-1]" sample for next iteration
goertzel_buf[i] = a; // save in "c" for subaudible tone detection
if (((!ads.fm_conf.squelched) && (!tone_det_enabled))
|| ((ads.fm_conf.subaudible_tone_detected) && (tone_det_enabled))
|| ((!ts.fm_sql_threshold)))// high-pass audio only if we are un-squelched (to save processor time)
{
// Do differentiating high-pass filter to attenuate very low frequency audio components, namely subadible tones and other "speaker-rattling" components - and to remove any DC that might be present.
float32_t b = FM_RX_HPF_ALPHA * (fm_data.hpf_prev_b + a - fm_data.hpf_prev_a);// do differentiation
fm_data.hpf_prev_a = a; // save "[n-1]" samples for next iteration
fm_data.hpf_prev_b = b;
a_buffer[i] = b;// save demodulated and filtered audio in main audio processing buffer
}
else if ((ads.fm_conf.squelched)
|| ((!ads.fm_conf.subaudible_tone_detected) && (tone_det_enabled)))// were we squelched or tone NOT detected?
{
a_buffer[i] = 0;// do not filter receive audio - fill buffer with zeroes to mute it
}
fm_data.q_prev = q_buffer[i];// save "previous" value of each channel to allow detection of the change of angle in next go-around
fm_data.i_prev = i_buffer[i];
}
// *** Squelch Processing ***
arm_iir_lattice_f32(&IIR_Squelch_HPF, squelch_buf, squelch_buf,
blockSize); // Do IIR high-pass filter on audio so we may detect squelch noise energy
ads.fm_conf.sql_avg = ((1 - FM_RX_SQL_SMOOTHING) * ads.fm_conf.sql_avg)
+ (FM_RX_SQL_SMOOTHING * sqrtf(fabsf(squelch_buf[0])));// IIR filter squelch energy magnitude: We need look at only one representative sample
//
// Squelch processing
//
// Determine if the (averaged) energy in "ads.fm.sql_avg" is above or below the squelch threshold
//
fm_data.count++;// bump count that controls how often the squelch threshold is checked
fm_data.count %= FM_SQUELCH_PROC_DECIMATION; // enforce the count limit
if (fm_data.count == 0) // do the squelch threshold calculation much less often than we are called to process this audio
{
if (ads.fm_conf.sql_avg > 0.175) // limit maximum noise value in averaging to keep it from going out into the weeds under no-signal conditions (higher = noisier)
{
ads.fm_conf.sql_avg = 0.175;
}
float32_t scaled_sql_avg = ads.fm_conf.sql_avg * 172;// scale noise amplitude to range of squelch setting
if (scaled_sql_avg > 24) // limit noise amplitude range
{
scaled_sql_avg = 24;
}
//
scaled_sql_avg = 22 - scaled_sql_avg;
// "invert" the noise power so that high number now corresponds with
// quieter signal: "scaled_sql_avg" may now be compared with squelch setting
// Now evaluate noise power with respect to squelch setting
if (ts.fm_sql_threshold == 0) // is squelch set to zero?
{
ads.fm_conf.squelched = false; // yes, the we are un-squelched
}
else
{
if (ads.fm_conf.squelched == true) // are we squelched?
{
if (scaled_sql_avg >= (float) (ts.fm_sql_threshold + FM_SQUELCH_HYSTERESIS)) // yes - is average above threshold plus hysteresis?
{
ads.fm_conf.squelched = false; // yes, open the squelch
}
}
else // is the squelch open (e.g. passing audio)?
{
if (ts.fm_sql_threshold > FM_SQUELCH_HYSTERESIS)// is setting higher than hysteresis?
{
if (scaled_sql_avg < (float) (ts.fm_sql_threshold - FM_SQUELCH_HYSTERESIS))// yes - is average below threshold minus hysteresis?
{
ads.fm_conf.squelched = true; // yes, close the squelch
}
}
else // setting is lower than hysteresis so we can't use it!
{
if (scaled_sql_avg < (float) ts.fm_sql_threshold)// yes - is average below threshold?
{
ads.fm_conf.squelched = true; // yes, close the squelch
}
}
}
}
}
//
// *** Subaudible tone detection ***
//
if (tone_det_enabled)// is subaudible tone detection enabled? If so, do decoding
{
//
// Use Goertzel algorithm for subaudible tone detection
//
// We will detect differentially at three frequencies: Above, below and on-frequency. The two former will be used to provide a sample of the total energy
// present as well as improve nearby-frequency discrimination. By dividing the on-frequency energy with the averaged off-frequency energy we'll
// get a ratio that is irrespective of the actual detected audio amplitude: A ratio of 1.00 is considered "neutral" and it goes above unity with the increasing
// likelihood that a tone was present on the target frequency
//
// Goertzel constants for the three decoders are pre-calculated in the function "UiCalcSubaudibleDetFreq()"
//
// (Yes, I know that below could be rewritten to be a bit more compact-looking, but it would not be much faster and it would be less-readable)
//
// Note that the "c" buffer contains audio that is somewhat low-pass filtered by the integrator, above
//
fm_data.gcount++;// this counter is used for the accumulation of data over multiple cycles
//
for (uint16_t i = 0; i < blockSize; i++)
{
// Detect above target frequency
AudioFilter_GoertzelInput(&ads.fm_conf.goertzel[FM_HIGH],goertzel_buf[i]);
// Detect energy below target frequency
AudioFilter_GoertzelInput(&ads.fm_conf.goertzel[FM_LOW],goertzel_buf[i]);
// Detect on-frequency energy
AudioFilter_GoertzelInput(&ads.fm_conf.goertzel[FM_CTR],goertzel_buf[i]);
}
if (fm_data.gcount >= FM_SUBAUDIBLE_GOERTZEL_WINDOW)// have we accumulated enough samples to do the final energy calculation?
{
float32_t s = AudioFilter_GoertzelEnergy(&ads.fm_conf.goertzel[FM_HIGH]) + AudioFilter_GoertzelEnergy(&ads.fm_conf.goertzel[FM_LOW]);
// sum +/- energy levels:
// s = "off frequency" energy reading
float32_t r = AudioFilter_GoertzelEnergy(&ads.fm_conf.goertzel[FM_CTR]);
fm_data.subdet = ((1 - FM_TONE_DETECT_ALPHA) * fm_data.subdet)
+ (r / (s / 2) * FM_TONE_DETECT_ALPHA); // do IIR filtering of the ratio between on and off-frequency energy
if (fm_data.subdet > FM_SUBAUDIBLE_TONE_DET_THRESHOLD)// is subaudible tone detector ratio above threshold?
{
fm_data.tdet++; // yes - increment count // yes - bump debounce count
if (fm_data.tdet > FM_SUBAUDIBLE_DEBOUNCE_MAX)// is count above the maximum?
{
fm_data.tdet = FM_SUBAUDIBLE_DEBOUNCE_MAX;// yes - limit the count
}
}
else // it is below the threshold - reduce the debounce
{
if (fm_data.tdet) // - but only if already nonzero!
{
fm_data.tdet--;
}
}
if (fm_data.tdet >= FM_SUBAUDIBLE_TONE_DEBOUNCE_THRESHOLD)// are we above the debounce threshold?
{
ads.fm_conf.subaudible_tone_detected = 1;// yes - a tone has been detected
}
else // not above threshold
{
ads.fm_conf.subaudible_tone_detected = 0; // no tone detected
}
fm_data.gcount = 0; // reset accumulation counter
}
}
else // subaudible tone detection disabled
{
ads.fm_conf.subaudible_tone_detected = 1;// always signal that a tone is being detected if detection is disabled to enable audio gate
}
}
return ads.fm_conf.squelched == false;
}
#if defined (USE_LMS_AUTONOTCH)
/**
* @author Clint, KA7OEI
*
* @param blockSize
* @param notchbuffer
*/
static void AudioDriver_NotchFilter(int16_t blockSize, float32_t *notchbuffer)
{
static ulong lms2_inbuf = 0;
static ulong lms2_outbuf = 0;
// DSP Automatic Notch Filter using LMS (Least Mean Squared) algorithm
//
arm_copy_f32(notchbuffer, &lmsData.lms2_nr_delay[lms2_inbuf], blockSize); // put new data into the delay buffer
//
arm_lms_norm_f32(&lmsData.lms2Norm_instance, notchbuffer, &lmsData.lms2_nr_delay[lms2_outbuf], lmsData.errsig2, notchbuffer, blockSize); // do automatic notch
// Desired (notched) audio comes from the "error" term - "errsig2" is used to hold the discarded ("non-error") audio data
//
lms2_inbuf += blockSize; // update circular de-correlation delay buffer
lms2_outbuf = lms2_inbuf + blockSize;
lms2_inbuf %= ts.dsp.notch_delaybuf_len;
lms2_outbuf %= ts.dsp.notch_delaybuf_len;
//
}
#endif
static void AudioDriver_Mix(float32_t* src, float32_t* dst, float32_t scaling, const uint16_t blockSize)
{
float32_t e3_buffer[blockSize];
arm_scale_f32(src, scaling, e3_buffer, blockSize);
arm_add_f32(dst, e3_buffer, dst, blockSize);
}
void AudioDriver_IQPhaseAdjust(uint16_t txrx_mode, float32_t* i_buffer, float32_t* q_buffer, const uint16_t blockSize)
{
int16_t trans_idx;
// right now only in TX used, may change in future
// if ((txrx_mode == TRX_MODE_TX && ts.dmod_mode == DEMOD_CW) || ts.iq_freq_mode == FREQ_IQ_CONV_MODE_OFF) // Comm. by UB8JDC
if ((!(ts.expflags1 & EXPFLAGS1_CLEAR_TX_CW_RTTY_BPSK) && (txrx_mode == TRX_MODE_TX && ts.dmod_mode == DEMOD_CW)) || ts.iq_freq_mode == FREQ_IQ_CONV_MODE_OFF) // UB8JDC
{
trans_idx = IQ_TRANS_OFF;
}
else
{
trans_idx = IQ_TRANS_ON;
}
float32_t iq_phase_balance = (txrx_mode == TRX_MODE_RX)? ads.iq_phase_balance_rx: ads.iq_phase_balance_tx[trans_idx];
if (iq_phase_balance < 0) // we only need to deal with I and put a little bit of it into Q
{
AudioDriver_Mix(i_buffer,q_buffer, iq_phase_balance, blockSize);
}
else if (iq_phase_balance > 0) // we only need to deal with Q and put a little bit of it into I
{
AudioDriver_Mix(q_buffer,i_buffer, iq_phase_balance, blockSize);
}
}
/**
* Takes an iq_buffer and puts the samples into a single interleaved spectrum ringbuffer
* To be called from the audio interrupt side of things. Do not call otherwise as the code is not
* reentrant will screw up things.
* @param iq_puf_b pair of i/q buffers
* @param blockSize how many samples in a single iq buffer, target gets 2*blockSize samples
*/
static void AudioDriver_SpectrumCopyIqBuffers(iq_buffer_t* iq_puf_b, const size_t blockSize)
{
for(int i = 0; i < blockSize; i++)
{
// Collect I/Q samples
sd.FFT_RingBuffer[sd.samp_ptr] = iq_puf_b->q_buffer[i]; // get floating point data for FFT for spectrum scope/waterfall display
sd.samp_ptr++;
sd.FFT_RingBuffer[sd.samp_ptr] = iq_puf_b->i_buffer[i];
sd.samp_ptr++;
if(sd.samp_ptr >= sd.fft_iq_len-1)
{
sd.samp_ptr = 0;
}
}
}
/**
* Places IQ data in the spectrums IQ ringbuffer if no zoom level has been set. Otherwise does nothing
* To be called from the audio interrupt side of things. Do not call otherwise as the code is not
* reentrant and may screw up things.
* Will place blockSize / 2^sd.magnify samples in ring buffer
*
* @param iq_buf_p iq data before frequency shift at IQ_SAMEPLE_RATE
* @param blockSize IQ sample count
*/
static void AudioDriver_SpectrumNoZoomProcessSamples(iq_buffer_t* iq_puf_b, const uint16_t blockSize)
{
if(sd.reading_ringbuffer == false && sd.fft_iq_len > 0)
{
if(sd.magnify == 0) //
{
AudioDriver_SpectrumCopyIqBuffers(iq_puf_b, blockSize);
sd.FFT_frequency = (ts.tune_freq); // spectrum shows all, LO is center frequency;
}
}
}
/**
* Places IQ data in the spectrums IQ ringbuffer if a zoom level has been set. Otherwise does nothing
* To be called from the audio interrupt side of things. Do not call otherwise as the code is not
* reentrant will screw up things.
* Will place blockSize / 2^sd.magnify samples in ring buffer
*
* @param iq_buf_p iq data with frequency of interest centered! (after frequency shift) at IQ_SAMEPLE_RATE
* @param blockSize IQ sample count
*/
static void AudioDriver_SpectrumZoomProcessSamples(iq_buffer_t* iq_buf_p, const uint16_t blockSize)
{
if(sd.reading_ringbuffer == false && sd.fft_iq_len > 0)
{
if(sd.magnify != 0) //
// magnify 2, 4, 8, 16, or 32
{
// ZOOM FFT
// is used here to have a very close look at a small part
// of the spectrum display of the mcHF
// The ZOOM FFT is based on the principles described in Lyons (2011)
// 1. take the I & Q samples
// 2. complex conversion to baseband (at this place has already been done in audio_rx_freq_conv!)
// 3. lowpass I and lowpass Q separately (48kHz sample rate)
// 4. decimate I and Q separately
// 5. apply 256-point-FFT to decimated I&Q samples
//
// frequency resolution: spectrum bandwidth / 256
// example: decimate by 8 --> 48kHz / 8 = 6kHz spectrum display bandwidth
// frequency resolution of the display --> 6kHz / 256 = 23.44Hz
// in 32x Mag-mode the resolution is 5.9Hz, not bad for such a small processor . . .
//
iq_buffer_t decim_iq_buf;
// lowpass Filtering
// Mag 2x - 12k lowpass --> 24k bandwidth
// Mag 4x - 6k lowpass --> 12k bandwidth
// Mag 8x - 3k lowpass --> 6k bandwidth
// Mag 16x - 1k5 lowpass --> 3k bandwidth
// Mag 32x - 750Hz lowpass --> 1k5 bandwidth
// 1st attempt - use low processor power biquad lowpass filters with 4 stages each
arm_biquad_cascade_df1_f32 (&IIR_biquad_Zoom_FFT_I, iq_buf_p->i_buffer,decim_iq_buf.i_buffer, blockSize);
arm_biquad_cascade_df1_f32 (&IIR_biquad_Zoom_FFT_Q, iq_buf_p->q_buffer,decim_iq_buf.q_buffer, blockSize);
// decimation
arm_fir_decimate_f32(&DECIMATE_ZOOM_FFT_I, decim_iq_buf.i_buffer, decim_iq_buf.i_buffer, blockSize);
arm_fir_decimate_f32(&DECIMATE_ZOOM_FFT_Q, decim_iq_buf.q_buffer, decim_iq_buf.q_buffer, blockSize);
// collect samples for spectrum display 256-point-FFT
AudioDriver_SpectrumCopyIqBuffers(&decim_iq_buf, blockSize/ (1<<sd.magnify));
// sd.FFT_frequency = ts.tune_freq + AudioDriver_GetTranslateFreq(); // spectrum shows center at translate frequency, LO + Translate Freq is center frequency;
if (ts.iq_freq_mode == FREQ_IQ_CONV_SLIDE)
{
sd.FFT_frequency = ts.tune_freq;
}
else
{
sd.FFT_frequency = ts.tune_freq + AudioDriver_GetTranslateFreq();
}
// TODO: also insert sample collection for snap carrier here
// and subsequently rebuild snap carrier to use a much smaller FFT (say 256 or 512)
// in order to save many kilobytes of RAM ;-)
}
}
}
static float32_t AudioDriver_FadeLeveler(int chan, float32_t audio, float32_t corr)
{
assert (chan < NUM_AUDIO_CHANNELS);
static float32_t dc27[NUM_AUDIO_CHANNELS]; // static will be initialized with 0
static float32_t dc_insert[NUM_AUDIO_CHANNELS];
dc27[chan] = adb.sam.mtauR * dc27[chan] + adb.sam.onem_mtauR * audio;
dc_insert[chan] = adb.sam.mtauI * dc_insert[chan] + adb.sam.onem_mtauI * corr;
audio = audio + dc_insert[chan] - dc27[chan];
return audio;
}
typedef struct
{
const float32_t c0[SAM_PLL_HILBERT_STAGES]; // Filter coefficients - path 0
const float32_t c1[SAM_PLL_HILBERT_STAGES]; // Filter coefficients - path 1
} demod_sam_const_t;
//sideband separation, these values never change
const demod_sam_const_t demod_sam_const =
{
.c0 = {
-0.328201924180698,
-0.744171491539427,
-0.923022915444215,
-0.978490468768238,
-0.994128272402075,
-0.998458978159551,
-0.999790306259206,
},
.c1 = {
-0.0991227952747244,
-0.565619728761389,
-0.857467122550052,
-0.959123933111275,
-0.988739372718090,
-0.996959189310611,
-0.999282492800792,
},
};
typedef struct
{
uint16_t count;
float32_t fil_out;
float32_t lowpass;
float32_t omega2;
float32_t phs;
float32_t dsI; // delayed sample, I path
float32_t dsQ; // delayed sample, Q path
#define OUT_IDX (3 * SAM_PLL_HILBERT_STAGES)
float32_t a[OUT_IDX + 3]; // Filter a variables
float32_t b[OUT_IDX + 3]; // Filter b variables
float32_t c[OUT_IDX + 3]; // Filter c variables
float32_t d[OUT_IDX + 3]; // Filter d variables
} demod_sam_data_t;
demod_sam_data_t sam_data;
/**
* Demodulate IQ carrying AM into audio, expects input to be at decimated input rate.
* Can do real synchronous AM demodulation with phase detector and PLL
*
* @author DD4WH
* @date December 2016
*
* @param i_buffer
* @param q_buffer
* @param a_buffer
* @param blockSize
*/
static void AudioDriver_DemodSAM(float32_t* i_buffer, float32_t* q_buffer, float32_t a_buffer[][IQ_BLOCK_SIZE], int16_t blockSize, float32_t sampleRate)
{
// new synchronous AM PLL & PHASE detector
// wdsp Warren Pratt, 2016
//*****************************
switch(ts.dmod_mode)
{
/*
* DSB (double sideband mode) if anybody needs that
*
// DSB = LSB + USB = (I - Q) + (I + Q)
case DEMOD_DSB:
arm_sub_f32(i_buffer, q_buffer, a_buffer[0], blockSize);
arm_add_f32(i_buffer, a_buffer[0], a_buffer[0], blockSize);
arm_add_f32(q_buffer, a_buffer[0], a_buffer[0], blockSize);
break;
*/
case DEMOD_AM:
for(int i = 0; i < blockSize; i++)
{
float32_t audio;
arm_sqrt_f32 (i_buffer[i] * i_buffer[i] + q_buffer[i] * q_buffer[i], &audio);
if(ads.fade_leveler)
{
audio = AudioDriver_FadeLeveler(0,audio,0);
}
a_buffer[0][i] = audio;
}
break;
case DEMOD_SAM:
{
// Wheatley 2011 cuteSDR & Warren Pratts WDSP, 2016
for(int i = 0; i < blockSize; i++)
{ // NCO
float32_t ai, bi, aq, bq;
float32_t Sin, Cos;
sincosf(sam_data.phs,&Sin,&Cos);
ai = Cos * i_buffer[i];
bi = Sin * i_buffer[i];
aq = Cos * q_buffer[i];
bq = Sin * q_buffer[i];
float32_t audio[NUM_AUDIO_CHANNELS];
// we initialize the often unused stereo channel
// to keep the compiler happy
#ifdef USE_TWO_CHANNEL_AUDIO
audio[1] = 0;
#endif
float32_t corr[2] = { ai + bq, -bi + aq };
if (ads.sam_sideband != SAM_SIDEBAND_BOTH)
{
sam_data.a[0] = sam_data.dsI;
sam_data.b[0] = bi;
sam_data.c[0] = sam_data.dsQ;
sam_data.d[0] = aq;
sam_data.dsI = ai;
sam_data.dsQ = bq;
for (int j = 0; j < SAM_PLL_HILBERT_STAGES; j++)
{
int k = 3 * j;
sam_data.a[k + 3] = demod_sam_const.c0[j] * (sam_data.a[k] - sam_data.a[k + 5]) + sam_data.a[k + 2];
sam_data.b[k + 3] = demod_sam_const.c1[j] * (sam_data.b[k] - sam_data.b[k + 5]) + sam_data.b[k + 2];
sam_data.c[k + 3] = demod_sam_const.c0[j] * (sam_data.c[k] - sam_data.c[k + 5]) + sam_data.c[k + 2];
sam_data.d[k + 3] = demod_sam_const.c1[j] * (sam_data.d[k] - sam_data.d[k + 5]) + sam_data.d[k + 2];
}
float32_t ai_ps = sam_data.a[OUT_IDX];
float32_t bi_ps = sam_data.b[OUT_IDX];
float32_t bq_ps = sam_data.c[OUT_IDX];
float32_t aq_ps = sam_data.d[OUT_IDX];
// make room for next sample
for (int j = OUT_IDX + 2; j > 0; j--)
{
sam_data.a[j] = sam_data.a[j - 1];
sam_data.b[j] = sam_data.b[j - 1];
sam_data.c[j] = sam_data.c[j - 1];
sam_data.d[j] = sam_data.d[j - 1];
}
switch(ads.sam_sideband)
{
default:
case SAM_SIDEBAND_USB:
audio[0] = (ai_ps - bi_ps) + (aq_ps + bq_ps);
break;
case SAM_SIDEBAND_LSB:
audio[0] = (ai_ps + bi_ps) - (aq_ps - bq_ps);
break;
#ifdef USE_TWO_CHANNEL_AUDIO
case SAM_SIDEBAND_STEREO:
audio[0] = (ai_ps + bi_ps) - (aq_ps - bq_ps);
audio[1] = (ai_ps - bi_ps) + (aq_ps + bq_ps);
break;
#endif
}
}
else
{
audio[0] = corr[0];
}
// "fade leveler", taken from Warren Pratts WDSP / HPSDR, 2016
// http://svn.tapr.org/repos_sdr_hpsdr/trunk/W5WC/PowerSDR_HPSDR_mRX_PS/Source/wdsp/
if(ads.fade_leveler)
{
for (int chan = 0; chan < NUM_AUDIO_CHANNELS; chan++)
{
audio[chan] = AudioDriver_FadeLeveler(chan, audio[chan], corr[0]);
}
}
/*
a->dc = a->mtauR * a->dc + a->onem_mtauR * audio;
a->dc_insert = a->mtauI * a->dc_insert + a->onem_mtauI * corr[0];
audio += a->dc_insert - a->dc;
*/
for (int chan = 0; chan < NUM_AUDIO_CHANNELS; chan++)
{
a_buffer[chan][i] = audio[chan];
}
// determine phase error
float32_t phzerror = atan2f(corr[1], corr[0]);
float32_t del_out = sam_data.fil_out;
// correct frequency 1st step
sam_data.omega2 = sam_data.omega2 + adb.sam.g2 * phzerror;
if (sam_data.omega2 < adb.sam.omega_min)
{
sam_data.omega2 = adb.sam.omega_min;
}
else if (sam_data.omega2 > adb.sam.omega_max)
{
sam_data.omega2 = adb.sam.omega_max;
}
// correct frequency 2nd step
sam_data.fil_out = adb.sam.g1 * phzerror + sam_data.omega2;
sam_data.phs = sam_data.phs + del_out;
// wrap round 2PI, modulus
while (sam_data.phs >= 2.0 * PI) { sam_data.phs -= (2.0 * PI); }
while (sam_data.phs < 0.0) { sam_data.phs += (2.0 * PI); }
}
sam_data.count++;
if(sam_data.count > 50) // to display the exact carrier frequency that the PLL is tuned to
// in the small frequency display
// we calculate carrier offset here and the display function is
// then called in UiDriver_MainHandler approx. every 40-80ms
{ // to make this smoother, a simple lowpass/exponential averager here . . .
float32_t carrier = 0.1 * (sam_data.omega2 * sampleRate) / (2.0 * PI);
carrier = carrier + 0.9 * sam_data.lowpass;
ads.carrier_freq_offset = carrier;
sam_data.count = 0;
sam_data.lowpass = carrier;
}
}
break;
}
}
/**
* Runs the interrupt part of the twin peaks detection and triggers codec restart
*
* @param iq_corr_p pointer to the previously calculated correction data
*/
static void AudioDriver_RxHandleTwinpeaks(const iq_correction_data_t* iq_corr_p)
{
// Test and fix of the "twinpeak syndrome"
// which occurs sporadically and can -to our knowledge- only be fixed
// by a reset of the codec
// It can be identified by a totally non-existing mirror rejection,
// so I & Q have essentially the same phase
// We use this to identify the snydrome and reset the codec accordingly:
// calculate phase between I & Q
static uint32_t twinpeaks_counter = 0;
static uint32_t codec_restarts = 0;
static float32_t phase_IQ; // used to determine the twin peaks issue state
static uint32_t phase_IQ_runs;
if (ts.twinpeaks_tested == TWINPEAKS_WAIT)
{
twinpeaks_counter++;
}
if(twinpeaks_counter > 1000) // wait 0.667s for the system to settle: with 32 IQ samples per block and 48ksps (0.66667ms/block)
{
ts.twinpeaks_tested = TWINPEAKS_SAMPLING;
twinpeaks_counter = 0;
phase_IQ = 0.0;
phase_IQ_runs = 0;
}
if(iq_corr_p->teta3 != 0.0 && ts.twinpeaks_tested == TWINPEAKS_SAMPLING) // prevent divide-by-zero
// twinpeak_tested = 2 --> wait for system to warm up
// twinpeak_tested = 0 --> go and test the IQ phase
// twinpeak_tested = 1 --> tested, verified, go and have a nice day!
// twinpeak_tested = 3 --> was not correctable
// twinpeak_tested = 4 --> we request the main loop to run the codec restart
// we can't do the restart inside the audio interrupt
{ // Moseley & Slump (2006) eq. (33)
// this gives us the phase error between I & Q in radians
float32_t phase_IQ_cur = asinf(iq_corr_p->teta1 / iq_corr_p->teta3);
// we combine 50 cycles (1/30s) to calculate the "final" phase_IQ
if (phase_IQ_runs == 0)
{
phase_IQ = phase_IQ_cur;
}
else
{
phase_IQ = 0.05 * phase_IQ_cur + 0.95 * phase_IQ;
}
phase_IQ_runs ++;
if (phase_IQ_runs == 50)
{
if (fabsf(phase_IQ) > (M_PI/8.0))
// threshold of 22.5 degrees phase shift == PI / 8
// hopefully your hardware is not so bad, that its phase error is more than 22 degrees ;-)
// if it is that bad, adjust this threshold to maybe PI / 7 or PI / 6
{
ts.twinpeaks_tested = TWINPEAKS_CODEC_RESTART;
codec_restarts++;
if(codec_restarts >= 4)
{
ts.twinpeaks_tested = TWINPEAKS_UNCORRECTABLE;
codec_restarts = 0;
}
}
else
{
ts.twinpeaks_tested = TWINPEAKS_DONE;
codec_restarts = 0;
}
}
}
}
/**
*
* @param blockSize
*/
static void AudioDriver_RxHandleIqCorrection(float32_t* i_buffer, float32_t* q_buffer, const uint16_t blockSize)
{
assert(blockSize >= 8);
if(!ts.iq_auto_correction) // Manual IQ imbalance correction
{
// Apply I/Q amplitude correction
arm_scale_f32(i_buffer, ts.rx_adj_gain_var.i, i_buffer, blockSize);
arm_scale_f32(q_buffer, ts.rx_adj_gain_var.q, q_buffer, blockSize); // TODO: we need only scale one channel! DD4WH, Dec 2016
// Apply I/Q phase correction
AudioDriver_IQPhaseAdjust(ts.txrx_mode, i_buffer, q_buffer, blockSize);
}
else // Automatic IQ imbalance correction
{ // Moseley, N.A. & C.H. Slump (2006): A low-complexity feed-forward I/Q imbalance compensation algorithm.
// in 17th Annual Workshop on Circuits, Nov. 2006, pp. 158-164.
// http://doc.utwente.nl/66726/1/moseley.pdf
for(uint32_t i = 0; i < blockSize; i++)
{
adb.iq_corr.teta1 += Math_sign_new(i_buffer[i]) * q_buffer[i]; // eq (34)
adb.iq_corr.teta2 += Math_sign_new(i_buffer[i]) * i_buffer[i]; // eq (35)
adb.iq_corr.teta3 += Math_sign_new(q_buffer[i]) * q_buffer[i]; // eq (36)
}
adb.iq_corr.teta1 = -0.003 * (adb.iq_corr.teta1 / blockSize) + 0.997 * adb.iq_corr.teta1_old; // eq (34) and first order lowpass
adb.iq_corr.teta2 = 0.003 * (adb.iq_corr.teta2 / blockSize) + 0.997 * adb.iq_corr.teta2_old; // eq (35) and first order lowpass
adb.iq_corr.teta3 = 0.003 * (adb.iq_corr.teta3 / blockSize) + 0.997 * adb.iq_corr.teta3_old; // eq (36) and first order lowpass
adb.iq_corr.M_c1 = (adb.iq_corr.teta2 != 0.0) ? adb.iq_corr.teta1 / adb.iq_corr.teta2 : 0.0; // eq (30)
// prevent divide-by-zero
float32_t help = (adb.iq_corr.teta2 * adb.iq_corr.teta2);
if(help > 0.0)// prevent divide-by-zero
{
help = (adb.iq_corr.teta3 * adb.iq_corr.teta3 - adb.iq_corr.teta1 * adb.iq_corr.teta1) / help; // eq (31)
}
adb.iq_corr.M_c2 = (help > 0.0) ? sqrtf(help) : 1.0; // eq (31)
// prevent sqrtf of negative value
AudioDriver_RxHandleTwinpeaks(&adb.iq_corr);
adb.iq_corr.teta1_old = adb.iq_corr.teta1;
adb.iq_corr.teta2_old = adb.iq_corr.teta2;
adb.iq_corr.teta3_old = adb.iq_corr.teta3;
adb.iq_corr.teta1 = 0.0;
adb.iq_corr.teta2 = 0.0;
adb.iq_corr.teta3 = 0.0;
// first correct Q and then correct I --> this order is crucially important!
for(uint32_t i = 0; i < blockSize; i++)
{ // see fig. 5
q_buffer[i] += adb.iq_corr.M_c1 * i_buffer[i];
}
// see fig. 5
arm_scale_f32 (i_buffer, adb.iq_corr.M_c2, i_buffer, blockSize);
}
}
/**
* This function is the interrupt side interface of the Noise Reduction
* @param blockSizeDecim
* @param inout_buffer the buffer carrying both the input data (subject to processing) and the processed
* output samples. Since the noise reduction uses a large buffer to be run on bigger chunks of samples
* there is a several milliseconds delay between the input being passed here and being returned processed
* Due to the buffering this function itself just puts one block in the buffer and returns a buffer if available
* or silence if not.
*/
static void AudioDriver_RxProcessorNoiseReduction(uint16_t blockSizeDecim, float32_t* inout_buffer)
{
#ifdef USE_ALTERNATE_NR
// we check if we can safely use the buffer, if not, we just do nothing
// TODO: add a better way to control which functions are used (available) in which mode
#ifdef USE_FREEDV
if ((ts.dmod_mode == DEMOD_DIGI && ts.digital_mode == DigitalMode_FreeDV) == false)
#endif
{
static int trans_count_in=0;
static int outbuff_count=0;
static int NR_fill_in_pt=0;
static NR_Buffer* out_buffer = NULL;
// this would be the right place for another decimation-by-2 to get down to 6ksps
// in order to further improve the spectral noise reduction
// anti-alias-filtering is already provided at this stage by the IIR main filter
// Only allow another decimation-by-two, if filter bandwidth is <= 2k7
//
// Add decimation-by-two HERE
// decide whether filter < 2k7!!!
uint32_t no_dec_samples = blockSizeDecim; // only used for the noise reduction decimation-by-two handling
// buffer needs max blockSizeDecim , less if second decimation is done
// see below
nr_params.NR_decimation_active = nr_params.NR_decimation_enable && (FilterInfo[ts.filters_p->id].width < 2701);
if (nr_params.NR_decimation_active == true)
{
no_dec_samples = blockSizeDecim / 2;
// decimate-by-2, DECIMATE_NR, in place
arm_fir_decimate_f32(&DECIMATE_NR, inout_buffer, inout_buffer, blockSizeDecim);
}
// attention -> change loop no into no_dec_samples!
for (int k = 0; k < no_dec_samples; k=k+2) //transfer our noisy audio to our NR-input buffer
{
mmb.nr_audio_buff[NR_fill_in_pt].samples[trans_count_in].real=inout_buffer[k];
mmb.nr_audio_buff[NR_fill_in_pt].samples[trans_count_in].imag=inout_buffer[k+1];
trans_count_in++; // count the samples towards FFT-size - 2 samples per loop
}
if (trans_count_in >= (NR_FFT_SIZE/2))
//NR_FFT_SIZE has to be an integer mult. of blockSizeDecim!!!
{
NR_in_buffer_add(&mmb.nr_audio_buff[NR_fill_in_pt]); // save pointer to full buffer
trans_count_in=0; // set counter to 0
NR_fill_in_pt++; // increase pointer index
NR_fill_in_pt %= NR_BUFFER_NUM; // make sure, that index stays in range
//at this point we have transfered one complete block of 128 (?) samples to one buffer
}
//**********************************************************************************
//don't worry! in the mean time the noise reduction routine is (hopefully) doing it's job within ui
//as soon as "fdv_audio_has_data" we can start harvesting the output
//**********************************************************************************
if (out_buffer == NULL && NR_out_has_data() > 1)
{
NR_out_buffer_peek(&out_buffer);
}
float32_t NR_dec_buffer[no_dec_samples];
if (out_buffer != NULL) //NR-routine has finished it's job
{
for (int j=0; j < no_dec_samples; j=j+2) // transfer noise reduced data back to our buffer
// for (int j=0; j < blockSizeDecim; j=j+2) // transfer noise reduced data back to our buffer
{
NR_dec_buffer[j] = out_buffer->samples[outbuff_count+NR_FFT_SIZE].real; //here add the offset in the buffer
NR_dec_buffer[j+1] = out_buffer->samples[outbuff_count+NR_FFT_SIZE].imag; //here add the offset in the buffer
outbuff_count++;
}
if (outbuff_count >= (NR_FFT_SIZE/2)) // we reached the end of the buffer coming from NR
{
outbuff_count = 0;
NR_out_buffer_remove(&out_buffer);
out_buffer = NULL;
NR_out_buffer_peek(&out_buffer);
}
}
else
{
memset(NR_dec_buffer,0,sizeof(NR_dec_buffer));
}
// interpolation of a_buffer from 6ksps to 12ksps!
// from NR_dec_buffer --> a_buffer
// but only, if we have decimated to 6ksps, otherwise just copy the samples into a_buffer
if (nr_params.NR_decimation_active == true)
{
arm_fir_interpolate_f32(&INTERPOLATE_NR, NR_dec_buffer, inout_buffer, no_dec_samples);
arm_scale_f32(inout_buffer, 2.0, inout_buffer, blockSizeDecim);
}
else
{
arm_copy_f32(NR_dec_buffer, inout_buffer, blockSizeDecim);
}
}
#endif
}
static void RxProcessor_DemodAudioPostprocessing(float32_t (*a_buffer)[AUDIO_BLOCK_SIZE], const size_t blockSize, const size_t blockSizeDecim, const uint32_t sampleRateDecim, const bool use_stereo)
{
const uint8_t dsp_active = ts.dsp.active;
const uint8_t dmod_mode = ts.dmod_mode;
if (ts.dsp.inhibit == false)
{
if((dsp_active & DSP_NOTCH_ENABLE) && (dmod_mode != DEMOD_CW) && !(dmod_mode == DEMOD_SAM && sampleRateDecim == 24000)) // No notch in CW
{
//#ifdef USE_LEAKY_LMS
// if(ts.enable_leaky_LMS)
// {
// AudioDriver_LeakyLmsNr(a_buffer[0], a_buffer[0], blockSizeDecim, 1);
// }
// else
//#endif
// commented by UB8JDC 2020.07.22
{
#ifdef USE_LMS_AUTONOTCH
AudioDriver_NotchFilter(blockSizeDecim, a_buffer[0]); // Do notch filter
#endif
}
}
#if defined(USE_LEAKY_LMS)
// DSP noise reduction using LMS (Least Mean Squared) algorithm
// This is the pre-filter/AGC instance
if((dsp_active & DSP_ANR_ENABLE) && (!(dsp_active & DSP_NR_POSTAGC_ENABLE)) && !(dmod_mode == DEMOD_SAM && sampleRateDecim == 24000)) // Do this if enabled and "Pre-AGC" DSP NR enabled
{
// if(ts.enable_leaky_LMS)
if(is_dsp_anr())
{
AudioDriver_LeakyLmsNr(a_buffer[0], a_buffer[0], blockSizeDecim, 0);
}
}
#endif
}
// Apply audio bandpass filter
if (IIR_PreFilter[0].numStages > 0) // yes, we want an audio IIR filter
{
arm_iir_lattice_f32(&IIR_PreFilter[0], a_buffer[0], a_buffer[0], blockSizeDecim);
#ifdef USE_TWO_CHANNEL_AUDIO
if(use_stereo)
{
arm_iir_lattice_f32(&IIR_PreFilter[1], a_buffer[1], a_buffer[1], blockSizeDecim);
}
#endif
}
// now process the samples and perform the receiver AGC function
AudioAgc_RunAgcWdsp(blockSizeDecim, a_buffer, use_stereo);
// DSP noise reduction using LMS (Least Mean Squared) algorithm
// This is the post-filter, post-AGC instance
#if defined(USE_LEAKY_LMS)
if((dsp_active & DSP_ANR_ENABLE) && (is_dsp_nr_postagc()) && (!ts.dsp.inhibit) && !(dmod_mode == DEMOD_SAM && sampleRateDecim == 24000)) // Do DSP NR if enabled and if post-DSP NR enabled
{
// if(ts.enable_leaky_LMS)
if(is_dsp_anr())
{
AudioDriver_LeakyLmsNr(a_buffer[0], a_buffer[0], blockSizeDecim, 0);
}
}
#endif
if (sampleRateDecim == 12000 && (is_dsp_nb_active() || is_dsp_nr())) //start of new nb or new noise reduction
{
// NR_in and _out buffers are using the same physical space than the freedv_iq_buffer in a
// shared MultiModeBuffer union.
// for NR reduction we use a maximum of 256 real samples
// so we use the freedv_iq buffers in a way, that we use the first half of each array for the input
// and the second half for the output
// .real and .imag are loosing there meaning here as they represent consecutive real samples
AudioDriver_RxProcessorNoiseReduction(blockSizeDecim, a_buffer[0]);
} // end of new nb
// Calculate scaling based on decimation rate since this affects the audio gain
const float32_t post_agc_gain_scaling =
(ts.filters_p->sample_rate_dec == RX_DECIMATION_RATE_12KHZ) ?
POST_AGC_GAIN_SCALING_DECIMATE_4 : POST_AGC_GAIN_SCALING_DECIMATE_2;
// Scale audio according to AGC setting, demodulation mode and required fixed levels and scaling
const float32_t scale_gain = post_agc_gain_scaling *
((dmod_mode == DEMOD_AM || dmod_mode == DEMOD_SAM) ?
0.5 /* AM/SAM */ : 0.333 /* not AM/SAM */);
// apply fixed amount of audio gain scaling to make the audio levels correct along with AGC
arm_scale_f32(a_buffer[0],scale_gain, a_buffer[0], blockSizeDecim);
// this is the biquad filter, a notch, peak, and lowshelf filter
arm_biquad_cascade_df1_f32 (&IIR_biquad_1[0], a_buffer[0],a_buffer[0], blockSizeDecim);
#ifdef USE_TWO_CHANNEL_AUDIO
if(use_stereo)
{
arm_scale_f32(a_buffer[1],scale_gain, a_buffer[1], blockSizeDecim); // apply fixed amount of audio gain scaling to make the audio levels correct along with AGC
arm_biquad_cascade_df1_f32 (&IIR_biquad_1[1], a_buffer[1],a_buffer[1], blockSizeDecim);
}
#endif
// all of these modems only work with 12 khz Samplerate
// TODO: User needs feedback if the modem is not activated due to wrong decimation rate
if (sampleRateDecim == 12000)
{
#ifdef USE_RTTY_PROCESSOR
if (is_demod_rtty()) // only works when decimation rate is 4 --> sample rate == 12ksps
{
AudioDriver_RxProcessor_Rtty(a_buffer[0], blockSizeDecim);
}
#endif
if (is_demod_psk()) // only works when decimation rate is 4 --> sample rate == 12ksps
{
AudioDriver_RxProcessor_Bpsk(a_buffer[0], blockSizeDecim);
}
if((dmod_mode == DEMOD_CW || dmod_mode == DEMOD_AM || dmod_mode == DEMOD_SAM))
// switch to use TUNE HELPER in AM/SAM
{
CwDecode_RxProcessor(a_buffer[0], blockSizeDecim);
}
}
// resample back to original sample rate while doing low-pass filtering to minimize audible aliasing effects
if (INTERPOLATE_RX[0].phaseLength > 0)
{
#ifdef USE_TWO_CHANNEL_AUDIO
float32_t temp_buffer[IQ_BLOCK_SIZE];
if(use_stereo)
{
arm_fir_interpolate_f32(&INTERPOLATE_RX[1], a_buffer[1], temp_buffer, blockSizeDecim);
}
#endif
arm_fir_interpolate_f32(&INTERPOLATE_RX[0], a_buffer[0], a_buffer[1], blockSizeDecim);
#ifdef USE_TWO_CHANNEL_AUDIO
if(use_stereo)
{
arm_copy_f32(temp_buffer,a_buffer[0],blockSize);
}
#endif
}
// additional antialias filter for specific bandwidths
// IIR ARMA-type lattice filter
if (IIR_AntiAlias[0].numStages > 0) // yes, we want an interpolation IIR filter
{
arm_iir_lattice_f32(&IIR_AntiAlias[0], a_buffer[1], a_buffer[1], blockSize);
#ifdef USE_TWO_CHANNEL_AUDIO
if(use_stereo)
{
arm_iir_lattice_f32(&IIR_AntiAlias[1], a_buffer[0], a_buffer[0], blockSize);
}
#endif
}
}
/**
* Gets IQ data as input, runs the rx processing on the input signal, leaves audio data in DMA buffer
*
* @param src iq input DMA buffer
* @param dst audio output DMA buffer
* @param blockSize number of input and output samples
* @param external_mute request to produce silence
*/
static void AudioDriver_RxProcessor(IqSample_t * const srcCodec, AudioSample_t * const dst, const uint16_t blockSize, bool external_mute)
{
// this is the main RX audio function
// it is driven with 32 samples in the complex buffer scr, meaning 32 * I AND 32 * Q
// blockSize is thus 32, DD4WH 2018_02_06
// we copy volatile variables which are used multiple times to local consts to let the compiler to its optimization magic
// since we are in an interrupt, no one will change these anyway
// shaved off a few bytes of code
const uint8_t dmod_mode = ts.dmod_mode;
const uint8_t tx_audio_source = ts.tx_audio_source;
const uint8_t rx_iq_source = ts.rx_iq_source;
const int32_t iq_freq_mode = ts.iq_freq_mode;
#ifdef USE_TWO_CHANNEL_AUDIO
const bool use_stereo = ((dmod_mode == DEMOD_IQ || dmod_mode == DEMOD_SSBSTEREO || (dmod_mode == DEMOD_SAM && ads.sam_sideband == SAM_SIDEBAND_STEREO)) && ts.stereo_enable);
#else
const bool use_stereo = false;
#endif
IqSample_t srcUSB[blockSize];
IqSample_t * const src = (rx_iq_source == RX_IQ_DIG || rx_iq_source == RX_IQ_DIGIQ) ? srcUSB : srcCodec;
// If source is digital usb in, pull from USB buffer, discard codec iq and
// let the normal processing happen
if (rx_iq_source == RX_IQ_DIG || rx_iq_source == RX_IQ_DIGIQ)
{
// audio sample rate must match the sample rate of USB audio if we read from USB
assert(rx_iq_source != RX_IQ_DIG || AUDIO_SAMPLE_RATE == USBD_AUDIO_FREQ);
// iq sample rate must match the sample rate of USB IQ audio if we read from USB
assert(tx_audio_source != RX_IQ_DIGIQ || IQ_SAMPLE_RATE == USBD_AUDIO_FREQ);
UsbdAudio_FillTxBuffer((AudioSample_t*)srcUSB,blockSize);
}
if (tx_audio_source == TX_AUDIO_DIGIQ)
{
for(uint32_t i = 0; i < blockSize; i++)
{
// 16 bit format - convert to float and increment
// we collect our I/Q samples for USB transmission if TX_AUDIO_DIGIQ
UsbdAudio_PutSample(I2S_IqSample_2_Int16(src[i].l));
UsbdAudio_PutSample(I2S_IqSample_2_Int16(src[i].r));
}
}
bool signal_active = false; // tells us if the modulator produced audio to listen to.
// if the audio filters are being reconfigured, we don't process audio at all
if (ads.af_disabled == 0 )
{
#ifdef USE_OBSOLETE_NOISEBLANKER
AudioDriver_NoiseBlanker(src, blockSize); // do noise blanker function
#endif
for(uint32_t i = 0; i < blockSize; i++)
{
int32_t level = abs(I2S_correctHalfWord(src[i].l))>>IQ_BIT_SHIFT;
if(level > ADC_CLIP_WARN_THRESHOLD/4) // This is the release threshold for the auto RF gain
{
ads.adc_quarter_clip = 1;
if(level > ADC_CLIP_WARN_THRESHOLD/2) // This is the trigger threshold for the auto RF gain
{
ads.adc_half_clip = 1;
if(level > ADC_CLIP_WARN_THRESHOLD) // This is the threshold for the red clip indicator on S-meter
{
ads.adc_clip = 1;
}
}
}
adb.iq_buf.i_buffer[i] = I2S_correctHalfWord(src[i].l);
adb.iq_buf.q_buffer[i] = I2S_correctHalfWord(src[i].r);
}
if (IQ_BIT_SCALE_DOWN != 1.0)
{
if(ts.codecCS4270_present)
{
// bla-bla-bla for CS4270 codec scaling for 24 bits
arm_scale_f32 (adb.iq_buf.i_buffer, IQ_BIT_SCALE_DOWN24, adb.iq_buf.i_buffer, blockSize);
arm_scale_f32 (adb.iq_buf.q_buffer, IQ_BIT_SCALE_DOWN24, adb.iq_buf.q_buffer, blockSize);
}
else
{
// we scale everything into the range of +/-32767 if we are getting 32 bit input
arm_scale_f32 (adb.iq_buf.i_buffer, IQ_BIT_SCALE_DOWN, adb.iq_buf.i_buffer, blockSize);
arm_scale_f32 (adb.iq_buf.q_buffer, IQ_BIT_SCALE_DOWN, adb.iq_buf.q_buffer, blockSize);
}
}
AudioDriver_RxHandleIqCorrection(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, blockSize);
// at this point we have phase corrected IQ @ IQ_SAMPLE_RATE, unshifted in adb.iq_buf.i_buffer, adb.iq_buf.q_buffer
// Spectrum display sample collect for magnify == 0
AudioDriver_SpectrumNoZoomProcessSamples(&adb.iq_buf, blockSize);
if(iq_freq_mode) // is receive frequency conversion to be done?
{
// FreqShift(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, blockSize, AudioDriver_GetTranslateFreq());
if (ts.iq_freq_mode == FREQ_IQ_CONV_SLIDE)
{
AudioDriver_SpectrumZoomProcessSamples(&adb.iq_buf, blockSize);
FreqShift(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, blockSize, AudioDriver_GetTranslateFreq());
}
else
{
FreqShift(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, blockSize, AudioDriver_GetTranslateFreq());
AudioDriver_SpectrumZoomProcessSamples(&adb.iq_buf, blockSize);
}
}
else
{
AudioDriver_SpectrumZoomProcessSamples(&adb.iq_buf, blockSize);
}
// at this point we have phase corrected IQ @ IQ_SAMPLE_RATE, with our RX frequency in the center (i.e. at 0 Hertz Shift)
// in adb.iq_buf.i_buffer, adb.iq_buf.q_buffer
// Spectrum display sample collect for magnify != 0
// AudioDriver_SpectrumZoomProcessSamples(&adb.iq_buf, blockSize);
#ifdef USE_FREEDV
if (ts.dvmode == true && ts.digital_mode == DigitalMode_FreeDV)
{
signal_active = AudioDriver_RxProcessorFreeDV(&adb.iq_buf, adb.a_buffer[1], blockSize);
}
#endif
if (signal_active == false)
{
signal_active = true;
// by default assume the modulator returns a signal, most do, except FM which may be squelched
// and sets this to false
const bool use_decimatedIQ =
((ts.filters_p->FIR_I_coeff_file == i_rx_new_coeffs) // lower than 3k8 bandwidth: new filters with excellent sideband suppression
&& dmod_mode != DEMOD_FM ) || dmod_mode == DEMOD_SAM || dmod_mode == DEMOD_AM ;
const int16_t blockSizeDecim = blockSize/ads.decimation_rate;
const uint16_t blockSizeIQ = use_decimatedIQ? blockSizeDecim: blockSize;
const uint32_t sampleRateIQ = use_decimatedIQ? ads.decimated_freq: IQ_SAMPLE_RATE;
// in some case we decimate the IQ before passing to demodulator, in some don't
// in any case, we do audio filtering on decimated data
// ------------------------
// In SSB and CW - Do 0-90 degree Phase-added Hilbert Transform
// In AM and SAM, the FIR below does ONLY low-pass filtering appropriate for the filter bandwidth selected, in
// which case there is ***NO*** audio phase shift applied to the I/Q channels.
/*
* AM -> wants decimated IQ input, (I and Q is shifted as part of decimation?)
* FM -> wants full IQ_SAMPLE_RATE input, shifted I and Q shifted by 90 degrees
* SSB-> wants for narrower bandwidth decimated IQ input, shifted I and Q shifted by 90 degrees
* SSB-> wants for wider bandwidth full IQ_SAMPLE_RATE input, shifted I and Q shifted by 90 degrees
*/
if(use_decimatedIQ)
{
arm_fir_decimate_f32(&DECIMATE_RX_I, adb.iq_buf.i_buffer, adb.iq_buf.i_buffer, blockSize); // LPF built into decimation (Yes, you can decimate-in-place!)
arm_fir_decimate_f32(&DECIMATE_RX_Q, adb.iq_buf.q_buffer, adb.iq_buf.q_buffer, blockSize); // LPF built into decimation (Yes, you can decimate-in-place!)
}
if(dmod_mode != DEMOD_SAM && dmod_mode != DEMOD_AM) // for SAM & AM leave out this processor-intense filter
{
// SECOND: Hilbert transform (for all but AM/SAM)
arm_fir_f32(&Fir_Rx_Hilbert_I,adb.iq_buf.i_buffer, adb.iq_buf.i_buffer, blockSizeIQ); // Hilbert lowpass +45 degrees
arm_fir_f32(&Fir_Rx_Hilbert_Q,adb.iq_buf.q_buffer, adb.iq_buf.q_buffer, blockSizeIQ); // Hilbert lowpass -45 degrees
}
// at this point we have (low pass filtered/decimated?) IQ, with our RX frequency in the center (i.e. at 0 Hertz Shift)
// in adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, block size is in blockSizeIQ
if (RadioManagement_UsesBothSidebands(dmod_mode) || dmod_mode == DEMOD_SAM )
{
// we must go here in DEMOD_SAM even if we effectively output only a single sideband
switch(dmod_mode)
{
case DEMOD_AM:
case DEMOD_SAM:
AudioDriver_DemodSAM(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, adb.a_buffer, blockSizeIQ, sampleRateIQ); // lowpass filtering, decimation, and SAM demodulation
break;
case DEMOD_FM:
signal_active = AudioDriver_DemodFM(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, adb.a_buffer[0], blockSize);
break;
#ifdef USE_TWO_CHANNEL_AUDIO
case DEMOD_IQ: // leave I & Q as they are!
arm_copy_f32(adb.iq_buf.i_buffer, adb.a_buffer[0], blockSizeIQ);
arm_copy_f32(adb.iq_buf.q_buffer, adb.a_buffer[1], blockSizeIQ);
break;
case DEMOD_SSBSTEREO: // LSB-left, USB-right
arm_add_f32(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, adb.a_buffer[0], blockSizeIQ); // sum of I and Q - USB
arm_sub_f32(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, adb.a_buffer[1], blockSizeIQ); // difference of I and Q - LSB
break;
#endif
}
}
else if (RadioManagement_LSBActive(dmod_mode))
{
// all LSB modes are demodulated the same way, we handed the special case DEMOD_SAM / SAM-L earlier
arm_sub_f32(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, adb.a_buffer[0], blockSizeIQ); // difference of I and Q - LSB
}
else // USB
{
// all USB modes are demodulated the same way, we handed the special case DEMOD_SAM / SAM-U earlier
arm_add_f32(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, adb.a_buffer[0], blockSizeIQ); // sum of I and Q - USB
}
// at this point we have our demodulated audio signal in adb.a_buffer[0]
// it may or may not need decimation before we go on with filtering the
// audio signal
if(dmod_mode != DEMOD_FM) // are we NOT in FM mode?
{
// If we are not, do decimation if not already done, filtering, DSP notch/noise reduction, etc.
if (use_decimatedIQ == false) // we did not already decimate the input earlier
{
// TODO HILBERT
arm_fir_decimate_f32(&DECIMATE_RX_I, adb.a_buffer[0], adb.a_buffer[0], blockSizeIQ); // LPF built into decimation (Yes, you can decimate-in-place!)
#ifdef USE_TWO_CHANNEL_AUDIO
if(use_stereo)
{
arm_fir_decimate_f32(&DECIMATE_RX_Q, adb.a_buffer[1], adb.a_buffer[1], blockSizeIQ); // LPF built into decimation (Yes, you can decimate-in-place!)
}
#endif
}
// at this point we are at the decimated audio sample rate
// we support multiple rates
// here also the various digital mode modems are called
RxProcessor_DemodAudioPostprocessing(adb.a_buffer, blockSize, blockSizeDecim, ads.decimated_freq, use_stereo);
// we get back blockSize audio at full sample rate
} // end NOT in FM mode
else
{
// it is FM - we don't do any decimation, interpolation, filtering or any other processing - just rescale audio amplitude
// resulting audio signal is in adb.a_buffer[1]
arm_scale_f32(
adb.a_buffer[0],
RadioManagement_FmDevIs5khz() ? FM_RX_SCALING_5K : FM_RX_SCALING_2K5,
adb.a_buffer[1],
blockSize); // apply fixed amount of audio gain scaling to make the audio levels correct along with AGC
AudioAgc_RunAgcWdsp(blockSize, adb.a_buffer, false); // FM is not using stereo
}
// this is the biquad filter, a highshelf filter
arm_biquad_cascade_df1_f32 (&IIR_biquad_2[0], adb.a_buffer[1],adb.a_buffer[1], blockSize);
#ifdef USE_TWO_CHANNEL_AUDIO
if(use_stereo)
{
arm_biquad_cascade_df1_f32 (&IIR_biquad_2[1], adb.a_buffer[0],adb.a_buffer[0], blockSize);
}
#endif
}
}
// at this point we have audio at AUDIO_SAMPLE_RATE in our adb.a_buffer[1] (and [0] if we are in stereo)
// if signal_active is true and we're ready to send it out
bool do_mute_output = external_mute == true || (signal_active == false);
if (do_mute_output)
// fill audio buffers with zeroes if we are to mute the receiver completely while still processing data OR it is in FM and squelched
// or when filters are switched
{
arm_fill_f32(0, adb.a_buffer[0], blockSize);
arm_fill_f32(0, adb.a_buffer[1], blockSize);
}
else
{
#ifdef USE_TWO_CHANNEL_AUDIO
// BOTH CHANNELS "FIXED" GAIN as input for audio amp and headphones/lineout
// each output path has its own gain control.
// Do fixed scaling of audio for LINE OUT
arm_scale_f32(adb.a_buffer[1], LINE_OUT_SCALING_FACTOR, adb.a_buffer[1], blockSize);
if (use_stereo)
{
arm_scale_f32(adb.a_buffer[0], LINE_OUT_SCALING_FACTOR, adb.a_buffer[0], blockSize);
}
else
{
// we simply copy the data from the other channel
arm_copy_f32(adb.a_buffer[1], adb.a_buffer[0], blockSize);
}
#else
// VARIABLE LEVEL FOR SPEAKER
// LINE OUT (constant level)
arm_scale_f32(adb.a_buffer[1], LINE_OUT_SCALING_FACTOR, adb.a_buffer[0], blockSize); // Do fixed scaling of audio for LINE OUT and copy to "a" buffer in one operation
#endif
//
// AF gain in "ts.audio_gain-active"
// 0 - 16: via codec command
// 17 - 20: soft gain after decoder
//
#ifdef UI_BRD_MCHF
if(ts.rx_gain[RX_AUDIO_SPKR].value > CODEC_SPEAKER_MAX_VOLUME) // is volume control above highest hardware setting?
{
arm_scale_f32(adb.a_buffer[1], (float32_t)ts.rx_gain[RX_AUDIO_SPKR].active_value, adb.a_buffer[1], blockSize); // yes, do software volume control adjust on "b" buffer
}
#endif
}
float32_t usb_audio_gain = ts.rx_gain[RX_AUDIO_DIG].value/31.0;
// we may have to play a key beep. We apply it to the left and right channel in OVI40 (on mcHF etc on left only because it is the speakers channel, not line out)
if((ts.beep_timing > 0)) // is beep active?
{
#ifdef USE_TWO_CHANNEL_AUDIO
softdds_addSingleToneToTwobuffers(&ads.beep, adb.a_buffer[0],adb.a_buffer[1], blockSize, ads.beep_loudness_factor);
#else
softdds_addSingleTone(&ads.beep, adb.a_buffer[1], blockSize, ads.beep_loudness_factor);
#endif
}
// Transfer processed audio to DMA buffer
for(int i=0; i < blockSize; i++) // transfer to DMA buffer and do conversion to INT
{
if (do_mute_output)
{
dst[i].l = 0;
dst[i].r = 0;
}
else
{
dst[i].l = adb.a_buffer[1][i];
dst[i].r = adb.a_buffer[0][i];
// in case we have to scale up our values from 16 to 32 bit range. Yes, we don't use the lower bits from the float
// but that probably does not make a difference and this way it is faster.
// the halfword correction is required when we are running on a STM32F4 with 32bit IQ, which at the moment (!) implies a AUDIO_BIT_SHIFT
// so we are safe FOR NOW! When we adjust everything to 32bit, the correction has to remain but can be moved to the float to int conversion
if (AUDIO_BIT_SHIFT != 0)
{
dst[i].l = I2S_correctHalfWord(dst[i].l << AUDIO_BIT_SHIFT);
dst[i].r = I2S_correctHalfWord(dst[i].r << AUDIO_BIT_SHIFT);
}
}
// Unless this is DIGITAL I/Q Mode, we sent processed audio
if (tx_audio_source != TX_AUDIO_DIGIQ)
{
int16_t vals[2];
#ifdef USE_TWO_CHANNEL_AUDIO
vals[0] = adb.a_buffer[0][i] * usb_audio_gain;
vals[1] = adb.a_buffer[1][i] * usb_audio_gain;
#else
vals[0] = vals[1] = adb.a_buffer[0][i] * usb_audio_gain;
#endif
UsbdAudio_PutSample(vals[0]);
UsbdAudio_PutSample(vals[1]);
}
}
}
static void AudioDriver_AudioFillSilence(AudioSample_t *s, size_t size)
{
memset(s,0,size*sizeof(*s));
}
static void AudioDriver_IqFillSilence(IqSample_t *s, size_t size)
{
memset(s,0,size*sizeof(*s));
}
//*----------------------------------------------------------------------------
//* Function Name : I2S_RX_CallBack
//* Object :
//* Object : audio sample processor
//* Input Parameters :
//* Output Parameters :
//* Functions called :
//*----------------------------------------------------------------------------
void AudioDriver_I2SCallback(AudioSample_t *audio, IqSample_t *iq, AudioSample_t *audioDst, int16_t blockSize)
{
static bool to_rx = false; // used as a flag to clear the RX buffer
static bool to_tx = false; // used as a flag to clear the TX buffer
bool muted = false;
if(ts.show_debug_info)
{
Board_GreenLed(LED_STATE_ON);
}
if((ts.txrx_mode == TRX_MODE_RX))
{
if((to_rx) || ts.audio_processor_input_mute_counter > 0) // the first time back to RX, clear the buffers to reduce the "crash"
{
muted = true;
AudioDriver_IqFillSilence(iq, blockSize);
if (to_rx)
{
UhsdrHwI2s_Codec_ClearTxDmaBuffer();
}
if ( ts.audio_processor_input_mute_counter >0)
{
ts.audio_processor_input_mute_counter--;
}
to_rx = false; // caused by the content of the buffers from TX - used on return from SSB TX
}
#ifdef SDR_AMBER
if(ts.codecCS4270_present && (ts.rf_gain_codecCS4270 != 1.0)) // simulation of level control at the ADC input, which is absent in the CS4270 codec
{
for(uint16_t i = 0; i < blockSize; i++)
{
iq[i].l = (uint32_t) (iq[i].l * ts.rf_gain_codecCS4270);
iq[i].r = (uint32_t) (iq[i].r * ts.rf_gain_codecCS4270);
}
}
#endif
#ifdef USE_CONVOLUTION
AudioDriver_RxProcessorConvolution(iq, audio, blockSize, muted);
#else
AudioDriver_RxProcessor(iq, audio, blockSize, muted);
#endif
if (ts.audio_dac_muting_buffer_count > 0)
{
ts.audio_dac_muting_buffer_count--;
}
if (muted == false)
{
if (ts.cw_keyer_mode != CW_KEYER_MODE_STRAIGHT && (ts.cw_text_entry || ts.dmod_mode == DEMOD_CW)) // FIXME to call always when straight mode reworked
{
CwGen_Process(adb.iq_buf.i_buffer, adb.iq_buf.q_buffer, blockSize);
}
}
to_tx = true; // Set flag to indicate that we WERE receiving when we go back to transmit mode
}
else // Transmit mode
{
if (to_tx)
{
TxProcessor_PrepareRun(); // last actions before we go live
}
if((to_tx) || (ts.audio_processor_input_mute_counter > 0) || ts.audio_dac_muting_flag || ts.audio_dac_muting_buffer_count > 0) // the first time back to TX, or TX audio muting timer still active - clear the buffers to reduce the "crash"
{
muted = true;
AudioDriver_AudioFillSilence(audio, blockSize);
to_tx = false; // caused by the content of the buffers from TX - used on return from SSB TX
if (ts.audio_processor_input_mute_counter > 0)
{
ts.audio_processor_input_mute_counter--;
}
}
TxProcessor_Run(audio, iq, audioDst,blockSize, muted);
// Pause or inactivity
if (ts.audio_dac_muting_buffer_count)
{
ts.audio_dac_muting_buffer_count--;
}
to_rx = true; // Set flag to indicate that we WERE transmitting when we eventually go back to receive mode
}
UiDriver_Callback_AudioISR();
if(ts.show_debug_info)
{
Board_GreenLed(LED_STATE_OFF);
}
#ifdef USE_PENDSV_FOR_HIGHPRIO_TASKS
// let us trigger a pendsv irq here in order to trigger execution of UiDriver_HighPrioHandler()
SCB->ICSR |= SCB_ICSR_PENDSVSET_Msk;
#endif
}
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