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Added radioModulatedGenerator_F32 for AM/PM/FM transmit

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pull/6/merge
boblark 4 years ago
parent f6e578b82c
commit 029b9b3769
3 changed files with 300 additions and 0 deletions
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  1. 1
      OpenAudio_ArduinoLibrary.h
  2. 124
      radioModulatedGenerator_F32.cpp
  3. 175
      radioModulatedGenerator_F32.h

1
OpenAudio_ArduinoLibrary.h
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#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"

124
radioModulatedGenerator_F32.cpp
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/* 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);
}

175
radioModulatedGenerator_F32.h
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/* 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
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