kopia lustrzana https://github.com/kamocat/uSDX
Refined SSB reception
Previous method was not correct - summing I and Q does not recover the original. Also removed os tests from main.cchibios
rodzic
3aed3f52f5
commit
67ef27a996
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@ -108,10 +108,6 @@ include $(CHIBIOS)/os/nil/nil.mk
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include $(CHIBIOS)/os/common/ports/ARMCMx/compilers/GCC/mk/port_v6m.mk
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# Auto-build files in ./source recursively.
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include $(CHIBIOS)/tools/mk/autobuild.mk
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# Other files (optional).
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include $(CHIBIOS)/test/lib/test.mk
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include $(CHIBIOS)/test/nil/nil_test.mk
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include $(CHIBIOS)/test/oslib/oslib_test.mk
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# Define linker script file here
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LDSCRIPT= $(STARTUPLD)/STM32F051x8.ld
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@ -119,7 +115,6 @@ LDSCRIPT= $(STARTUPLD)/STM32F051x8.ld
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# C sources that can be compiled in ARM or THUMB mode depending on the global
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# setting.
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CSRC = $(ALLCSRC) \
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$(TESTSRC) \
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main.c \
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radio/hilbert.c \
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radio/rx.c \
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@ -24,18 +24,48 @@ struct synth{
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uint8_t phase:7; //< Phase offset, in increments of 1/(vxco*4) seconds
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};
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/** Sets the PLL and Clock in the synth config
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*
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* Does not reset the clock setup
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*/
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void synthInit(struct synth * cfg, uint8_t channel, uint8_t pll);
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/** Starts the clock output
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*
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*/
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void synthStart(struct synth * cfg);
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/** Sets the clock to the desired frequency
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*
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* Calls synthWriteConfig internally
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*/
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void synthSetCarrier(struct synth * cfg, float carrier);
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/** Adjusts the clock by a certain frequency
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*
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* This is much faster than synthSetCarrier, but has limited range.
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* Also, the frequency may have error due to linearization.
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*
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* @param baseband 15.16 fixed-point value, in Hz
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*/
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void synthSetBaseband(struct synth * cfg, int32_t baseband);
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/** Writes a and b parameters for a pll multiplier or divider
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*/
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void synthWriteParam(uint8_t reg, uint64_t val, uint8_t div);
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/** Writes config to device
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*
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* This may be useful for restoring configurations without recalculating
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*/
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void synthWriteConfig(struct synth * cfg);
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/* Sets the initial phase offset
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/** Sets the initial phase offset
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*
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* Note that this has limited resolution and range. The maximum delay is 31.75x the PLL clock period,
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* which means you will have a difficult time getting a 90-degree phase shift on the 80m band.
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*/
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void synthSetPhase(struct synth * cfg, float degrees);
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/** Stops the clock output
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*
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* This may be helpful for power saving
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* Unless otherwise configured, clock output will be held low while stopped
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*/
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void synthStop(struct synth * cfg);
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@ -16,8 +16,6 @@
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#include "hal.h"
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#include "ch.h"
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#include "nil_test_root.h"
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#include "oslib_test_root.h"
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#include "drivers/si5351.h"
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static void adcerrorcallback(ADCDriver *adcp, adcerror_t err) {
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@ -133,12 +131,7 @@ THD_FUNCTION(Thread3, arg) {
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/* Welcome message.*/
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chnWrite(&SD1, (const uint8_t *)"Hello World!\r\n", 14);
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/* Waiting for encoder turn and activation of the test suite.*/
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while (true) {
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if (palReadLine(LINE_ENC0)) {
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test_execute((BaseSequentialStream *)&SD1, &nil_test_suite);
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test_execute((BaseSequentialStream *)&SD1, &oslib_test_suite);
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}
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chThdSleepMilliseconds(500);
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}
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}
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@ -7,7 +7,7 @@
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#include "hilbert.h"
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int16_t hilbert19(int16_t * src){
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int32_t hilbert19(int16_t * src){
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const size_t M = 19/2;
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/* Hilbert coeffecients with a hamming window
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* Values are 2/pi, 2/3pi, 2/7pi, 2/9pi, etc
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@ -20,5 +20,25 @@ int16_t hilbert19(int16_t * src){
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sum += (src[M-5]-src[M+5]) * (int32_t)coef[2];
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sum += (src[M-7]-src[M+7]) * (int32_t)coef[3];
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sum += (src[M-9]-src[M+9]) * (int32_t)coef[4];
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return sum>>16;
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return sum;
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}
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int32_t hilbert32(int16_t * src, uint8_t i){
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i -= 16; // Move to the center of the history
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const uint8_t mask = 32-1;
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/* Hilbert coeffecients with a hamming window
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* Values are 2/pi, 2/3pi, 2/7pi, 2/9pi, etc
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* Stored in 15.16 fixed-point
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*/
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const int16_t coef[] = {41353, 12829, 6638, 3753, 2087, 1079, 506, 247};
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int32_t sum;
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sum = (src[(i- 1)&mask]-src[(i+ 1)&mask]) * (int32_t)coef[0];
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sum += (src[(i- 3)&mask]-src[(i+ 3)&mask]) * (int32_t)coef[1];
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sum += (src[(i- 5)&mask]-src[(i+ 5)&mask]) * (int32_t)coef[2];
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sum += (src[(i- 7)&mask]-src[(i+ 7)&mask]) * (int32_t)coef[3];
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sum += (src[(i- 9)&mask]-src[(i+ 9)&mask]) * (int32_t)coef[4];
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sum += (src[(i-11)&mask]-src[(i+11)&mask]) * (int32_t)coef[5];
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sum += (src[(i-13)&mask]-src[(i+13)&mask]) * (int32_t)coef[6];
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sum += (src[(i-15)&mask]-src[(i+15)&mask]) * (int32_t)coef[7];
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return sum;
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}
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@ -6,4 +6,12 @@
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*
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* Generated with hamming window
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*/
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int16_t hilbert19(int16_t * src);
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int32_t hilbert19(int16_t * src);
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/** 31-element Hilbert transform
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*
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* This implementation assumes a 32-element buffer, and wraps accordingly
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* @param src The source data. Make sure I and Q are not interlieved
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* @param i The index of the latest element. Older elements are assumed to have decreasing index.
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*/
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int32_t hilbert32(int16_t * src, uint8_t i);
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@ -7,8 +7,8 @@
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#include "rx.h"
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#include "ssb.h"
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#include "../drivers/speaker.h"
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#include "../drivers/si5351.h"
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#include "../drivers/speaker.h"
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/** Mailbox for received data */
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mailbox_t new_sample;
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@ -18,23 +18,16 @@ struct{
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enum radio_mode mode;
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}rx_cfg;
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THD_WORKING_AREA(waradio_rx, 128);
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THD_WORKING_AREA(waradio_rx, 512);
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THD_FUNCTION(radio_rx, arg){
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(void)arg;
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const int len = 256;
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int16_t * data = chCoreAllocFromBase(len*sizeof(int16_t), sizeof(int16_t), 0);
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while(1){
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union complex c;
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for( int i=0; i<len;){
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chMBFetchTimeout(&new_sample, &c.msg, TIME_INFINITE);
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data[i++] = c.real;
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data[i++] = c.imag;
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}
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/** Process the received data */
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int16_t out[len];
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ssb_rx(out, data, len);
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/** Fill buffer for audio out */
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speakerUpdate(out, len);
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if((USB==rx_cfg.mode) || (LSB==rx_cfg.mode))
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ssb_rx(&new_sample, &(rx_cfg.mode));
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else if(CW==rx_cfg.mode) // Currently use SSB for CW decoding
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ssb_rx(&new_sample, &(rx_cfg.mode));
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else
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chThdSleepMilliseconds(50); // Don't hog all the CPU
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}
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}
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@ -46,7 +39,7 @@ THD_FUNCTION(radio_rx, arg){
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* We use a second-order IIR filter to prevent aliasing, followed
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* by an accumulate for decimation
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*/
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static void adccallback(ADCDriver *adcp) {
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void adccallback(ADCDriver *adcp) {
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adcsample_t * ptr = adcp->samples;
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size_t len = adcp->depth/2; // This also determines downsample ratio
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// These are persistent variables for filter history
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@ -77,7 +70,7 @@ static void adccallback(ADCDriver *adcp) {
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chMBPostTimeout(&new_sample, c.msg, TIME_IMMEDIATE);
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}
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static void adcerrorcallback(ADCDriver *adcp, adcerror_t err) {
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void adcerrorcallback(ADCDriver *adcp, adcerror_t err) {
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(void)adcp;
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(void)err;
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@ -144,6 +137,7 @@ void rxStart(enum radio_mode mode, float frequency){
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}
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void rxStop(void){
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rx_cfg.mode = STOPPED;
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adcStop(&ADCD1);
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speakerStop();
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}
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@ -16,6 +16,7 @@
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* This list will be expanded as more modes are supported
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*/
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enum radio_mode {
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STOPPED, //< Receiver is stopped
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CW, //< Continuous wave, also known as morse code
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USB,//< Upper Sideband, for frequencies above 10MHz
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LSB,//< Lower sideband, for frequencies below 10MHz
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@ -60,7 +61,7 @@ THD_FUNCTION(radio_rx, arg);
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* Performs initial low-pass filter and downsamples to 5kHz
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* Passes data to the Radio RX thread
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*/
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static void adccallback(ADCDriver *adcp);
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void adccallback(ADCDriver *adcp);
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/** ADC init
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*
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@ -5,15 +5,52 @@
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* Author: marshal
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*/
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#include "ssb.h"
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#include "hilbert.h"
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#include "../drivers/speaker.h"
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void ssb_rx(int16_t * dest, int16_t * src, size_t qty){
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// Source data is interlieved In-phase / Quadrature phase
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// Destination is single-channel
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for(int i = 0; i < qty; ++i){
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*dest++ = *src++ + *src++; // Return the sum of I and Q
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msg_t ssb_rx(mailbox_t * inbox, enum radio_mode * mode){
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union complex c;
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const int len = 32;
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int16_t real[len];
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int16_t imag[len];
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while((USB==*mode) || (LSB==*mode)){
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for( int i=0; i<len;){
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/** Fetch new data */
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msg_t m = chMBFetchTimeout(inbox, &c.msg, TIME_MS2I(10));
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if(MSG_OK != m)
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return m;
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real[i] = c.real;
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imag[i] = c.imag;
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if(LSB==*mode)
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imag[i]=-imag[i];
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/** Process the received data */
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int16_t out;
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uint8_t j = (i-16)&31; // get to the center of the data
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#ifdef HIGH_EMPH // Remove low frequencies
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int32_t h = hilbert32(imag, i);
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out = real[j] - (h>>16);
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#else // Keep low frequencies
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int32_t h = hilbert32(real, i);
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out = imag[j] + (h>>16);
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#endif
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/** Fill buffer for audio out
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* We could optionally feed more than one sample in at a time*/
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speakerUpdate(&out, 1);
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}
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}
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return MSG_OK;
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}
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const int32_t fscale = 1; //FIXME
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void ssb_tx(int16_t * amp, int32_t * freq, int16_t * src, size_t qty){
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// Source data is single-channel audio
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int32_t f = freq[0];
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for(size_t i = 0; i < (qty-18); ++i){
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amp[i]=src[i+9];
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// FIXME: Subtraction is not a differentiator
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// Maybe the differentiation can be combined with the hilbert?
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f -= hilbert19(src+i) * fscale;
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freq[i]=f;
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}
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}
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@ -6,11 +6,12 @@
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*/
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#include <stdint.h>
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#include <stddef.h>
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#include "rx.h"
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/** Single sideband decoder
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*
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*/
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void ssb_rx(int16_t * dest, int16_t * src, size_t qty);
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msg_t ssb_rx(mailbox_t * inbox, enum radio_mode * mode);
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/** Singgle sideband encoder
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*
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