/* * Copyright (C) 2020 by Jonathan Naylor G4KLX * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ #include "FMControl.h" #include #if defined(DUMP_RF_AUDIO) #include #endif const float EMPHASIS_GAIN_DB = 0.0F; //Gain needs to be the same for pre an deeemphasis const float FILTER_GAIN_DB = 0.0F; const unsigned int FM_MASK = 0x00000FFFU; CFMControl::CFMControl(CFMNetwork* network) : m_network(network), m_enabled(false), m_incomingRFAudio(1600U, "Incoming RF FM Audio"), m_preemphasis (NULL), m_deemphasis (NULL), m_filterStage1(NULL), m_filterStage2(NULL), m_filterStage3(NULL) { m_preemphasis = new CIIRDirectForm1Filter(0.38897032f, -0.32900053f, 0.0f, 1.0f, 0.28202918f, 0.0f, EMPHASIS_GAIN_DB); m_deemphasis = new CIIRDirectForm1Filter(1.0f,0.28202918f, 0.0f, 0.38897032f, -0.32900053f, 0.0f, EMPHASIS_GAIN_DB); //cheby type 1 0.2dB cheby type 1 3rd order 300-2700Hz fs=8000 m_filterStage1 = new CIIRDirectForm1Filter(0.29495028f, 0.0f, -0.29495028f, 1.0f, -0.61384624f, -0.057158668f, FILTER_GAIN_DB); m_filterStage2 = new CIIRDirectForm1Filter(1.0f, 2.0f, 1.0f, 1.0f, 0.9946123f, 0.6050482f, FILTER_GAIN_DB); m_filterStage3 = new CIIRDirectForm1Filter(1.0f, -2.0f, 1.0f, 1.0f, -1.8414584f, 0.8804949f, FILTER_GAIN_DB); } CFMControl::~CFMControl() { delete m_preemphasis ; delete m_deemphasis ; delete m_filterStage1; delete m_filterStage2; delete m_filterStage3; } bool CFMControl::writeModem(const unsigned char* data, unsigned int length) { assert(data != NULL); assert(length > 0U); if (m_network == NULL) return true; if (data[0U] == TAG_HEADER) return true; if (data[0U] == TAG_EOT) return m_network->writeEOT(); if (data[0U] != TAG_DATA) return false; m_incomingRFAudio.addData(data + 1U, length - 1U); unsigned int bufferLength = m_incomingRFAudio.dataSize(); if (bufferLength > 255U) bufferLength = 255U; if (bufferLength >= 3U) { #if defined(DUMP_RF_AUDIO) FILE * audiofile = fopen("./audiodump.bin", "ab"); #endif bufferLength = bufferLength - bufferLength % 3U; //round down to nearest multiple of 3 unsigned char bufferData[255U]; m_incomingRFAudio.getData(bufferData, bufferLength); unsigned int nSamples = 0; float samples[85U]; // 255 / 3; // Unpack the serial data into float values. for (unsigned int i = 0U; i < bufferLength; i += 3U) { short sample1 = 0U; short sample2 = 0U; unsigned int pack = 0U; unsigned char* packPointer = (unsigned char*)&pack; packPointer[0U] = bufferData[i]; packPointer[1U] = bufferData[i + 1U]; packPointer[2U] = bufferData[i + 2U]; //extract unsigned 12 bit samples to 16 bit signed sample2 = short(int(pack & FM_MASK) - 2048); sample1 = short(int(pack >> 12) - 2048); // Convert from unsigned short (0 - +4095) to float (-1.0 - +1.0) samples[nSamples++] = float(sample1) / 2048.0F; samples[nSamples++] = float(sample2) / 2048.0F; } //De-emphasise the data and remove CTCSS for (unsigned int i = 0U; i < nSamples; i++) { samples[i] = m_deemphasis->filter(samples[i]); samples[i] = m_filterStage3->filter(m_filterStage2->filter(m_filterStage1->filter(samples[i]))); } unsigned short out[170U]; // 85 * 2 unsigned int nOut = 0U; // Repack the data (8-bit unsigned values containing unsigned 16-bit data) for (unsigned int i = 0U; i < nSamples; i++) { unsigned short sample = (unsigned short)((samples[i] + 1.0F) * 32767.0F + 0.5F); out[nOut++] = ((sample >> 8) & 0x00FFU) | ((sample << 8) & 0xFF00U);//change endianess to network order, transmit MSB first } #if defined(DUMP_RF_AUDIO) if(audiofile != NULL) fwrite(out, sizeof(unsigned short), nOut, audiofile); #endif #if defined(DUMP_RF_AUDIO) if(audiofile != NULL) fclose(audiofile); #endif return m_network->writeData((unsigned char*)out, nOut * sizeof(unsigned short)); } return true; } unsigned int CFMControl::readModem(unsigned char* data, unsigned int space) { assert(data != NULL); assert(space > 0U); if (m_network == NULL) return 0U; if(space > 252U) space = 252U; unsigned char netData[168U];//84 * 2 modem can handle up to 84 samples (252 bytes) at a time unsigned int length = m_network->read(netData, 168U); if (length == 0U) return 0U; float samples[84U]; unsigned int nSamples = 0U; // Convert the unsigned 16-bit data (+65535 - 0) to float (+1.0 - -1.0) for (unsigned int i = 0U; i < length; i += 2U) { unsigned short sample = (netData[i + 0U] << 8) | netData[i + 1U]; samples[nSamples++] = (float(sample) / 32768.0F) - 1.0F; } //Pre-emphasise the data and other stuff. //for (unsigned int i = 0U; i < nSamples; i++) // samples[i] = m_preemphasis->filter(samples[i]); // Pack the floating point data (+1.0 to -1.0) to packed 12-bit samples (+2047 - -2048) unsigned int pack = 0U; unsigned char* packPointer = (unsigned char*)&pack; unsigned int j = 0U; unsigned int i = 0U; for (; i < nSamples && j < space; i += 2U, j += 3U) { unsigned short sample1 = (unsigned short)((samples[i] + 1.0F) * 2048.0F + 0.5F); unsigned short sample2 = (unsigned short)((samples[i + 1] + 1.0F) * 2048.0F + 0.5F); pack = 0U; pack = ((unsigned int)sample1) << 12; pack |= sample2; data[j] = packPointer[0U]; data[j + 1U] = packPointer[1U]; data[j + 2U] = packPointer[2U]; } return j;//return the number of bytes written } void CFMControl::clock(unsigned int ms) { // May not be needed } void CFMControl::enable(bool enabled) { // May not be needed }