kopia lustrzana https://github.com/ArjanteMarvelde/uSDR-pico
better sampling
ArjanteMarvelde
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151
dsp.c
151
dsp.c
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@ -10,10 +10,9 @@
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* This fifo is written from core0 from a 16us timer callback routine (i.e. 62.5kHz)
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*
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* The RX branch:
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* - Sample I and Q QSD channels intermittently, and shift into I and Q delay line (31.25 kHz per channel)
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* - Sample I and Q QSD channels, and shift into I and Q delay line (62.5 kHz per channel)
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* - Low pass filter: Fc=4kHz
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* - Interpolate last two I samples to correct sampling phase difference with Q
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* - Quarter rate (7.8125 kHz) to improve low F behavior of Hilbert transform
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* - Quarter rate (15.625 kHz) to improve low F behavior of Hilbert transform
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* - Calculate 15 tap Hilbert transform on Q
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* - Demodulate, SSB: Q - I, taking proper delays into account
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* - Push to Audio output DAC
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@ -41,8 +40,9 @@
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* DAC_RANGE defines PWM cycle, determining DAC resolution and PWM frequency.
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* DAC resolution = Vcc / DAC_RANGE
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* PWM frequency = Fsys / DAC_RANGE
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* A value of 500 means 125MHz/500=250kHz [or 250 and 500kHz]
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* A value of 250 means 125MHz/250=500kHz
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* ADC is 12 bit, so resolution is by definition 4096
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* To eliminate undefined behavior we clip off the upper 4 sample bits.
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*/
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#define DAC_RANGE 250
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#define DAC_BIAS DAC_RANGE/2
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@ -53,7 +53,7 @@
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* Callback timeout and inter-core FIFO commands.
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* The timer value in usec determines frequency of TX and RX loops
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* Exact time is obtained by passing the value as negative
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* Here we use 16us (62.5 kHz == PWM freq/4 [or 8])
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* Here we use 16us (62.5 kHz == PWM freq/4)
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*/
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#define DSP_US 16
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#define DSP_TX 1
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@ -84,101 +84,82 @@ void dsp_ptt(bool active)
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/*
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* CORE1:
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* Execute RX branch signal processing
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* No interleaving, since only 2us per conversion
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*/
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volatile int16_t i_s_pre[15], q_s_pre[15]; // Raw I/Q samples minus DC bias
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volatile int16_t i_s_raw[15], q_s_raw[15]; // Raw I/Q samples minus DC bias
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volatile int16_t i_s[15], q_s[15]; // Filtered I/Q samples
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volatile int16_t i_dc, q_dc; // DC bias for I/Q channel
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volatile int16_t i_prev; // Last I-channel raw sample
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volatile int dec_cnt=0; // Decimation counter
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bool rx(void)
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{
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static bool q_phase;
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int16_t sample;
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int32_t accu;
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int16_t q_sample, i_sample;
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int32_t q_accu, i_accu;
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int16_t qh;
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int i;
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if (q_phase)
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adc_select_input(1); // Q channel ADC
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q_sample = (int16_t)adc_read(); // Take Q sample
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adc_select_input(0); // I channel ADC
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i_sample = (int16_t)adc_read(); // Take I sample
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/*
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* Shift in I and Q raw samples
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*/
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for (i=0; i<14; i++)
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{
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adc_select_input(1); // Q channel ADC
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sample = (int16_t)adc_read() - ADC_BIAS; // Take sample and subtract mid-level
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/*
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* Shift Q samples
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*/
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for (i=0; i<14; i++)
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q_s_pre[i] = q_s_pre[i+1]; // Q samples delay line
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/*
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* Remove DC and store new sample
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* w(t) = x(t) + a*w(t-1) (use a=7/8, ca 0.87)
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* y(t) = w(t) - w(t-1)
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*/
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// sample += (((q_dc<<3)-q_dc)>>3); // Use sample as temporary q_dc
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// q_s_pre[14] = sample - q_dc; // Calculate output
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// q_dc = sample; // Store new q_dc
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q_s_pre[14] = sample;
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/*
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* Low pass filter
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*/
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for (i=0; i<14; i++) // Shift samples
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q_s[i] = q_s[i+1];
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accu = 0;
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for (i=0; i<15; i++) // Low pass FIR filter
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accu += (int32_t)q_s_pre[i]*lpf3_31[i]; // 3kHz, at 31.25 kHz sampling
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q_s[14] = accu / 256;
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q_phase = false; // Next: I branch
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q_s_raw[i] = q_s_raw[i+1]; // Q raw samples shift register
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i_s_raw[i] = i_s_raw[i+1]; // I raw samples shift register
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}
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else
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{
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adc_select_input(0); // I channel ADC
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sample = (int16_t)adc_read() - ADC_BIAS; // Take sample and subtract mid-level
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/*
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* Shift I samples
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*/
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for (i=0; i<14; i++)
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i_s_pre[i] = i_s_pre[i+1]; // I samples delay line
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/*
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* Remove DC and store new sample
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* w(t) = x(t) + a*w(t-1) (use a=7/8, ca 0.87)
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* y(t) = w(t) - w(t-1)
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*/
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// q_sample += (((q_dc<<3)-q_dc)>>3); // Use sample as temporary q_dc
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// q_s_raw[14] = q_sample - q_dc; // Calculate output
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// q_dc = q_sample; // Store new q_dc
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q_s_raw[14] = (q_sample&0x0fff) - ADC_BIAS; // just clip and subtract mid-level
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/*
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* Remove DC and store new sample: average last two to get in phase with Q
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* w(t) = x(t) + a*w(t-1) (use a=7/8, ca 0.87)
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* y(t) = w(t) - w(t-1)
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*/
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// sample += (((i_dc<<3)-i_dc)>>3); // Use sample as temporary i_dc
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// i_s_pre[14] = sample - i_dc; // Calculate output
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// i_dc = sample; // Store new i_dc
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// sample = i_s_pre[14]; // Get out uncorrected sample
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i_s_pre[14] = (sample + i_prev)/2; // Correct for phase difference with Q samples
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i_prev = sample; // Remember last sample for next I-phase
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/*
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* Low pass filter
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*/
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for (i=0; i<14; i++) // Shift samples
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i_s[i] = i_s[i+1];
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accu = 0;
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for (i=0; i<15; i++) // Low pass FIR filter
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accu += (int32_t)i_s_pre[i]*lpf3_31[i]; // 3kHz, at 31.25 kHz sampling
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i_s[14] = accu / 256;
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/*
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* Classic Hilbert transform 15 taps, 12 bits (see Iowa Hills):
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*/
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accu = (q_s[0]-q_s[14])*315L + (q_s[2]-q_s[12])*440L + (q_s[4]-q_s[10])*734L + (q_s[6]-q_s[ 8])*2202L;
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qh = accu / 4096;
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// -_sample += (((i_dc<<3)-i_dc)>>3); // Use sample as temporary i_dc
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// i_s_raw[14] = i_sample - i_dc; // Calculate output
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// i_dc = i_sample; // Store new i_dc
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i_s_raw[14] = (i_sample&0x0fff) - ADC_BIAS; // just clip and subtract mid-level
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/*
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* SSB demodulate: I[7] - Qh
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* Range should be within DAC_RANGE
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* Add 250 offset and send to audio DAC output
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*/
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sample = (i_s[7] - qh)/16;
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pwm_set_chan_level(dac_audio, PWM_CHAN_A, DAC_BIAS + sample);
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/*
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* Low pass filter + decimation
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*/
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dec_cnt = (dec_cnt+1)&0x03; // Use only every fourth sample
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if (dec_cnt>0) return true;
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q_phase = true; // Next: Q branch
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for (i=0; i<14; i++) // Shift samples
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{
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q_s[i] = q_s[i+1];
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i_s[i] = i_s[i+1];
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}
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q_accu = 0;
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i_accu = 0;
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for (i=0; i<15; i++) // Low pass FIR filter
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{
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q_accu += (int32_t)q_s_raw[i]*lpf3_62[i]; // Fc=3kHz, at 62.5 kHz sampling
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i_accu += (int32_t)i_s_raw[i]*lpf3_62[i]; // Fc=3kHz, at 62.5 kHz sampling
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}
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q_s[14] = q_accu >> 8;
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i_s[14] = q_accu >> 8;
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/*
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* Classic Hilbert transform 15 taps, 12 bits (see Iowa Hills)
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*/
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q_accu = (q_s[0]-q_s[14])*315L + (q_s[2]-q_s[12])*440L + (q_s[4]-q_s[10])*734L + (q_s[6]-q_s[ 8])*2202L;
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qh = q_accu >> 12;
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/*
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* SSB demodulate: I[7] - Qh
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* Range should be within DAC_RANGE
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* Add 250 offset and send to audio DAC output
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*/
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q_sample = (i_s[7] - qh)/8;
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pwm_set_chan_level(dac_audio, PWM_CHAN_A, DAC_BIAS + q_sample);
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return true;
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}
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