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FMControl.cpp
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FMControl.cpp
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/*
* Copyright (C) 2020,2021 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 <string>
#if defined(DUMP_RF_AUDIO)
#include <cstdio>
#endif
#define SWAP_BYTES_16(a) (((a >> 8) & 0x00FFU) | ((a << 8) & 0xFF00U))
const float DEEMPHASIS_GAIN_DB = 8.0F; // Audio gain adjustment
const float PREEMPHASIS_GAIN_DB = 0.0F; // Audio gain adjustment
const float FILTER_GAIN_DB = 2.0F; // Audio gain adjustment
const unsigned int FM_MASK = 0x00000FFFU;
CFMControl::CFMControl(CFMNetwork* network, float txAudioGain, float rxAudioGain, bool preEmphasisOn, bool deEmphasisOn) :
m_network(network),
m_txAudioGain(txAudioGain),
m_rxAudioGain(rxAudioGain),
m_preEmphasisOn(preEmphasisOn),
m_deEmphasisOn(deEmphasisOn),
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)
{
assert(txAudioGain > 0.0F);
assert(rxAudioGain > 0.0F);
m_preEmphasis = new CIIRDirectForm1Filter(8.315375384336983F, -7.03334621603483F,0.0F,1.0F, 0.282029168302153F,0.0F, PREEMPHASIS_GAIN_DB);
m_deEmphasis = new CIIRDirectForm1Filter(0.07708787090460224F, 0.07708787090460224F,0.0F, 1.0F, -0.8458242581907955F,0.0F, DEEMPHASIS_GAIN_DB);
// Chebyshev 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->writeEnd();
if (data[0U] != TAG_DATA)
return false;
m_incomingRFAudio.addData(data + 1U, length - 1U);
unsigned int bufferLength = m_incomingRFAudio.dataSize();
if (bufferLength > 240U) // 160 samples 12-bit
bufferLength = 240U; // 160 samples 12-bit
if (bufferLength >= 3U) {
bufferLength = bufferLength - bufferLength % 3U; // Round down to nearest multiple of 3
unsigned char bufferData[240U]; // 160 samples 12-bit
m_incomingRFAudio.getData(bufferData, bufferLength);
unsigned int pack = 0U;
unsigned char* packPointer = (unsigned char*)&pack;
float out[160U]; // 160 samples 12-bit
unsigned int nOut = 0U;
short unpackedSamples[2U];
for (unsigned int i = 0U; i < bufferLength; i += 3U) {
// Extract unsigned 12 bit unsigned sample pairs packed into 3 bytes to 16 bit signed
packPointer[0U] = bufferData[i + 0U];
packPointer[1U] = bufferData[i + 1U];
packPointer[2U] = bufferData[i + 2U];
unpackedSamples[1U] = short(int(pack & FM_MASK) - 2048);
unpackedSamples[0U] = short(int(pack >> 12 & FM_MASK) - 2048); //
// Process unpacked sample pair
for (unsigned char j = 0U; j < 2U; j++) {
// Convert to float (-1.0 to +1.0)
float sampleFloat = (float(unpackedSamples[j]) * m_rxAudioGain) / 2048.0F;
// De-emphasise and remove CTCSS
if (m_deEmphasisOn)
sampleFloat = m_deEmphasis->filter(sampleFloat);
out[nOut++] = m_filterStage3->filter(m_filterStage2->filter(m_filterStage1->filter(sampleFloat)));
}
}
#if defined(DUMP_RF_AUDIO)
FILE * audiofile = fopen("./audiodump.bin", "ab");
if (audiofile != NULL) {
fwrite(out, sizeof(float), nOut, audiofile);
fclose(audiofile);
}
#endif
return m_network->writeData(out, nOut);
}
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 > 240U) // 160 samples 12-bit
space = 240U; // 160 samples 12-bit
float netData[160U]; // Modem can handle up to 160 samples at a time
unsigned int length = m_network->readData(netData, 160U); // 160 samples 12-bit
if (length == 0U)
return 0U;
unsigned int pack = 0U;
unsigned char* packPointer = (unsigned char*)&pack;
unsigned int nData = 0U;
for (unsigned int i = 0; i < length; i++) {
float sampleFloat = netData[i] * m_txAudioGain;
// Pre-emphasis
if (m_preEmphasisOn)
sampleFloat = m_preEmphasis->filter(sampleFloat);
// Convert float to 12-bit samples (0 to 4095)
unsigned int sample12bit = (unsigned int)((sampleFloat + 1.0F) * 2048.0F + 0.5F);
// Pack 2 samples into 3 bytes
if ((i & 1U) == 0) {
pack = 0U;
pack = sample12bit << 12;
} else {
pack |= sample12bit;
data[nData++] = packPointer[0U];
data[nData++] = packPointer[1U];
data[nData++] = packPointer[2U];
}
}
return nData;
}
void CFMControl::clock(unsigned int ms)
{
// May not be needed
}
void CFMControl::enable(bool enabled)
{
// May not be needed
}