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