/* 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