/* * RadioIQMixer_F32.cpp * * 22 March 2020 * Bob Larkin, in support of the library: * Chip Audette, OpenAudio, Apr 2017 * ------------------- * A single signal channel comes in and is multiplied (mixed) with a sin * and cos of the same frequency. The pair of mixer outputs are * referred to as i and q. The conversion in frequency is either * up or down, and a pair of filters on i and q determine which is allow * to pass to the output. * * The sin/cos LO is from synth_sin_cos_f32.cpp See that for details. * * There are two then two outputs. * * MIT License, Use at your own risk. */ #include "RadioIQMixer_F32.h" // 513 values of the sine wave in a float array: #include "sinTable512_f32.h" void RadioIQMixer_F32::update(void) { audio_block_f32_t *blockIn, *blockOut_i=NULL, *blockOut_q=NULL; uint16_t index, i; float32_t a, b, deltaPhase, phaseC; // Get first input, i, that will be filtered blockIn = AudioStream_F32::receiveWritable_f32(0); if (!blockIn) { if(errorPrintIQM) Serial.println("IQMIXER-ERR: No input memory"); return; } // Try to get a pair of blocks for the IQ output blockOut_i = AudioStream_F32::allocate_f32(); if (!blockOut_i){ // Didn't have any if(errorPrintIQM) Serial.println("IQMIXER-ERR: No I output memory"); AudioStream_F32::release(blockIn); return; } blockOut_q = AudioStream_F32::allocate_f32(); if (!blockOut_q){ if(errorPrintIQM) Serial.println("IQMIXER-ERR: No Q output memory"); AudioStream_F32::release(blockIn); AudioStream_F32::release(blockOut_i); return; } // doSimple has amplitude (-1, 1) and sin/cos differ by 90.00 degrees. if (doSimple) { for (i=0; i < block_size; i++) { phaseS += phaseIncrement; if (phaseS > 512.0f) phaseS -= 512.0f; index = (uint16_t) phaseS; 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]; // Linear interpolation and multiplying (DBMixer) with input blockOut_i->data[i] = blockIn->data[i] * (a + 0.001953125*(b-a)*deltaPhase); /* Repeat for cosine by adding 90 degrees phase */ index = (index + 128) & 0x01ff; /* Read two nearest values of input value from the sin table */ a = sinTable512_f32[index]; b = sinTable512_f32[index+1]; /* deltaPhase will be the same as used for sin */ blockOut_q->data[i] = blockIn->data[i]*(a + 0.001953125*(b-a)*deltaPhase); } } else { // Do a more flexible update, i.e., not doSimple for (i=0; i < block_size; i++) { phaseS += phaseIncrement; if (phaseS > 512.0f) phaseS -= 512.0f; index = (uint16_t) phaseS; 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]; // We now have a sine value, so multiply with the input data and save // Linear interpolate sine and multiply with the input and amplitude (about 1.0) blockOut_i->data[i] = amplitude_pk * blockIn->data[i] * (a + 0.001953125*(b-a)*deltaPhase); /* Shift forward phaseS_C and get cos. First, the calculation of index of the table */ phaseC = phaseS + phaseS_C; if (phaseC > 512.0f) 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]; // Same as sin, but leave amplitude of LO at +/- 1.0 blockOut_q->data[i] = blockIn->data[i] * (a + 0.001953125*(b-a)*deltaPhase); } } AudioStream_F32::release(blockIn); // Done with this //transmit the data AudioStream_F32::transmit(blockOut_i, 0); // send the I outputs AudioStream_F32::release(blockOut_i); AudioStream_F32::transmit(blockOut_q, 1); // and the Q outputs AudioStream_F32::release(blockOut_q); }