Added radioModulatedGenerator_F32 for AM/PM/FM transmit

OpenAudio_ArduinoLibrary.h

@@ -40,6 +40,7 @@
 #include "FFT_Overlapped_OA_F32.h"
 #include "AudioEffectFreqShiftFD_OA_F32.h"
 #include "AudioEffectDelay_OA_F32.h"
+#include "radioModulatedGenerator_F32.h"
 #include "RadioIQMixer_F32.h"
 #include "AudioFilter90Deg_F32.h"
 #include "AudioAnalyzePhase_F32.h"

radioModulatedGenerator_F32.cpp

@@ -0,0 +1,124 @@
+/*     radioModulatedGenerator_F32.cpp
+ *
+ * RadioModulatedGenerator_F32 class - See .h file for information.
+ * Copyright (c) 2021 Bob Larkin            Created: 15 April 2021
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#include "radioModulatedGenerator_F32.h"
+
+// 513 values of the sine wave in a float array:
+#include "sinTable512_f32.h"
+
+void radioModulatedGenerator_F32::update(void) {
+  audio_block_f32_t *inAmpl,  *inPhaseFreq;
+  audio_block_f32_t *outBlockI, *outBlockQ;
+  uint16_t index, i;
+  float32_t a, b, deltaPhase, phaseC, amSig;
+
+
+ uint32_t tt=micros();
+
+  // Input 0 is for amplitude modulation.
+  if(doAM) {
+    inAmpl = AudioStream_F32::receiveReadOnly_f32(0);
+    if (!inAmpl)  return;
+    }
+
+  // Input 1 is for phase or frequency modulation.
+  if(doPM || doFM)  {
+    inPhaseFreq = AudioStream_F32::receiveReadOnly_f32(1);
+    if (!inPhaseFreq) {
+       if(doAM) AudioStream_F32::release(inAmpl);
+       return;
+    }
+  }
+
+  outBlockI = AudioStream_F32::allocate_f32();   // Output blocks
+  if (!outBlockI)  {
+     if(doAM) AudioStream_F32::release(inAmpl);
+     if(doPM || doFM) AudioStream_F32::release(inPhaseFreq);
+     return;
+  }
+
+  if(bothIQ)  {
+     outBlockQ = AudioStream_F32::allocate_f32();
+     if (!outBlockQ) {
+        if(doAM) AudioStream_F32::release(inAmpl);
+        if(doPM || doFM) AudioStream_F32::release(inPhaseFreq);
+        AudioStream_F32::release(outBlockI);
+        return;
+     }
+  }
+
+  for (i=0; i < block_length; i++) {
+     if(doPM)  // Phase in inPhaseFreq->data[i] is scaled for (0.0, 2*PI)
+        phaseS += (phaseIncrement0 + K512ON2PI*inPhaseFreq->data[i]);
+     else if(doFM)
+        phaseS += kp*(freq + inPhaseFreq->data[i]);  //  kp=512.0/sample_rate_Hz
+     else
+        phaseS += phaseIncrement0;  // No PM or FM alteration to carrier phase
+
+     while (phaseS > 512.0f)
+        phaseS -= 512.0f;
+     while (phaseS < 0.0f)
+        phaseS += 512.0f;
+     index = (uint16_t) phaseS;    // Does adding 0.5 here cut errors?  <<<<<<<<<<<<<<<<<<
+     deltaPhase = phaseS -(float32_t) index;
+     /* Read two nearest values of input value from the sin table */
+     a = sinTable512_f32[index];
+     b = sinTable512_f32[index+1];
+     if(doAM) {
+        amSig = 1.0f + inAmpl->data[i];
+        if(amSig<0.0f)
+           amSig = 0.0f;        // Common def of AM going back to vacuum tubes
+        outBlockI->data[i] = amplitude_pk*amSig*(a + 0.001953125*(b-a)*deltaPhase);  /* Linear interpolation process */
+     }
+     else
+        outBlockI->data[i] = amplitude_pk*(a + 0.001953125*(b-a)*deltaPhase);
+
+     if(bothIQ)  {
+        /* Shift forward phaseQ_I  and get cos. First, the calculation of index of the table */
+        phaseC = phaseS + phaseQ_I;
+        while (phaseC > 512.0f)
+           phaseC -= 512.0f;
+        while (phaseC < 0.0f)
+           phaseC += 512.0f;
+        index = (uint16_t) phaseC;
+        deltaPhase = phaseC -(float32_t) index;
+        /* Read two nearest values of input value from the sin table */
+        a = sinTable512_f32[index];
+        b = sinTable512_f32[index+1];
+        if(doAM)                  // amSig from above
+          outBlockQ->data[i] = amplitudeQ_I*amplitude_pk*amSig*(a + 0.001953125*(b-a)*deltaPhase);
+        else
+          outBlockQ->data[i] = amplitudeQ_I*amplitude_pk*(a + 0.001953125*(b-a)*deltaPhase);
+      }
+   }
+   if(doAM)         AudioStream_F32::release(inAmpl);
+   if(doPM || doFM) AudioStream_F32::release(inPhaseFreq);
+   AudioStream_F32::transmit(outBlockI, 0);
+   AudioStream_F32::release (outBlockI);
+   if(bothIQ)  {
+      AudioStream_F32::transmit(outBlockQ, 1);
+      AudioStream_F32::release (outBlockQ);
+   }
+Serial.println(micros() - tt);
+}

radioModulatedGenerator_F32.h

@@ -0,0 +1,175 @@
+/*     radioModulatedGenerator_F32.h
+ *
+ *   RadioModulatedGenerator_F32 class
+ *
+ * Created: Bob Larkin 15 April 2021
+ *
+ * For AM, the input is the 0 (left) channel. 100% AM modulation corresponds
+ * to this input -1.0 to 1.0.  Overmodulation (more that 100%) results in peak
+ * increases beyond twice amplitude, but full abrupt clipping at the
+ * bottom zero point.  Clipping on the top would be in an external block,
+ * if desired
+ *
+ * For PM or FM (only one at a time) the input goes to the 1 channel.  For PM,
+ * the input level corresponds to radians of phase change, + or -.  For FM,
+ * the input correspondss to Hz of deviation.
+ *
+ * For digital modulation, such as QAM, there can be both phase and amplitude
+ * modulation.  This would be set by
+ *      doModulation_AM_PM_FM(true, true, false, bool _bothIQ)
+ *
+ * If _bothIQ is false, the output is all at channel 0.  This is a standard
+ * modulated waveform as would be transmitted by wires or radio.  If _bothIQ
+ * is true, a pair of outputs on channels 0 and 1 correspond to I and Q
+ * components, as would be used with "phasing mixers" to convert the transmit
+ * frequency.
+ *
+ * Amplitude and phase corrections can be applied when there I-Q outputs.
+ * This can compensate for errors in the external hardware.  See the functions:
+ *      phaseQ_I(float32_t ph)
+ *      amplitudeQ_I(float32_t _a)
+ *
+ * Time: T3.6 update() block of 128 is about 53 microseconds  AM Single output
+ *       T4.x update() block of 128 is about 20 microseconds  AM Single output
+ *       T4.x update() block of 128 is about 35 microseconds  AM I + Q outputs
+ *       For T4.x, FM is 1 or 2 microseconds faster than AM.
+ *
+ * Copyright (c) 2021 Bob Larkin
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef modulate_AM_PM_FM_f32_h_
+#define modulate_AM_PM_FM_f32_h_
+
+#include "AudioStream_F32.h"
+#include "arm_math.h"
+
+#ifndef M_PI
+#define M_PI   3.14159265358979323846
+#endif
+
+#ifndef M_PI_2
+#define M_PI_2  1.57079632679489661923
+#endif
+
+#ifndef M_TWOPI
+#define M_TWOPI  (M_PI * 2.0)
+#endif
+
+#define MF2_PI 6.2831853f
+#define K512ON2PI 81.487331f
+
+class radioModulatedGenerator_F32 : public AudioStream_F32  {
+//GUI: inputs:2, outputs:2 //this line used for automatic generation of GUI node
+//GUI: shortName:Modulator  //this line used for automatic generation of GUI node
+public:
+    radioModulatedGenerator_F32(void) : AudioStream_F32(2, inputQueueArray_f32) { } //uses default AUDIO_SAMPLE_RATE from AudioStream.h
+    radioModulatedGenerator_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32) {
+        setSampleRate_Hz(settings.sample_rate_Hz);
+        setBlockLength(settings.audio_block_samples);
+    }
+
+    void frequency(float32_t fr) {    // Center Frequency in Hz
+        freq = fr;
+        if (freq < 0.0f) freq = 0.0f;
+        else if (freq > sample_rate_Hz/2.0f) freq = sample_rate_Hz/2.0f;
+        phaseIncrement0 = 512.0f * freq / sample_rate_Hz;
+    }
+
+    /* Externally, phase comes in the range (0,2*M_PI) keeping with C math functions
+     * Internally,  the full circle is represented as (0.0, 512.0).  This is
+     * convenient for finding the entry to the sine table.
+     */
+    void phase_r(float32_t ph) {
+        while (ph < 0.0f) ph += MF2_PI;
+        while (ph > MF2_PI) ph -= MF2_PI;
+        phaseS = 512.0f * ph / MF2_PI;
+        return;
+    }
+
+    // phaseQ_I  is the number of radians that the cosine output leads the
+    // sine output.  The default is M_PI_2 = pi/2 = 1.57079633 radians,
+    // corresponding to 90.00 degrees cosine leading sine.
+    void phaseQ_I_r(float32_t ph) {
+        while (ph < 0.0f) ph += MF2_PI;
+        while (ph > MF2_PI) ph -= MF2_PI;
+        // Internally a full circle is 512.00 of phase
+        phaseQ_I = 512.0f * ph / MF2_PI;
+        return;
+    }
+
+    // amplitudeQ_I  an amplitude unbalance introduced to the Q channel to
+    // compensate for errors in external hardware..
+    void amplitudeQI(float32_t _a) {
+        amplitudeQ_I = _a;
+        return;
+    }
+
+    // The amplitude, a, is the peak, as in zero-to-peak.  This produces outputs
+    // ranging from -a to +a.  Both outputs are the same amplitude.
+    // This will be multiplied by the AM input from Input 0.  This is "power control"
+    void amplitude(float32_t _a) {
+        amplitude_pk = _a;
+        return;
+        }
+
+    void doModulation_AM_PM_FM(bool _doAM, bool _doPM, bool _doFM, bool _bothIQ)  {
+        doAM = _doAM;
+        doPM = _doPM;
+        doFM = _doFM;
+        if(doPM & doFM)  doFM=false;  // One at a time
+        bothIQ = _bothIQ;
+        }
+
+    // Do not use.  For now, dynamic sample rate is not generally supported.
+    void setSampleRate_Hz(float32_t fs_Hz) {
+		sample_rate_Hz = fs_Hz;
+        // Check freq range
+        if (freq > sample_rate_Hz/2.0f) freq = sample_rate_Hz/2.0f;
+        // update phase increment for new frequency, and kp
+        phaseIncrement0 = 512.0f * freq/fs_Hz;
+        kp = 512.0f/sample_rate_Hz;
+        }
+
+    // Do not use.  Dynamic block length is un-supported.
+    void setBlockLength(uint16_t bl) {
+      if(bl > 128)  bl = 128;
+      block_length = bl;
+      }
+
+    virtual void update(void);
+
+private:
+    audio_block_f32_t *inputQueueArray_f32[2];
+    float32_t freq = 10000.0f;  // Center frequecy, Hz
+    float32_t phaseS = 0.0f;
+    float32_t phaseQ_I = 128.00;
+    float32_t amplitudeQ_I = 1.0f;
+    float32_t amplitude_pk = 1.0f;
+    float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE;  // Base, center freq
+    float32_t kp = 512.0/sample_rate_Hz;
+    float32_t phaseIncrement0 = kp*freq;;
+    uint16_t  block_length = 128;
+    bool      doAM = false;
+    bool      doPM = false;
+    bool      doFM = false;
+    bool      bothIQ = false;  // Quadrature outputs for analog mixers
+};
+#endif