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AMBE feature: creation and changes to DSD demod
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f4exb
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///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 F4EXB //
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// written by Edouard Griffiths //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// (at your option) any later version. //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#include <algorithm>
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#include <chrono>
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#include <thread>
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#include "audio/audiofifo.h"
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#include "ambeworker.h"
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MESSAGE_CLASS_DEFINITION(AMBEWorker::MsgMbeDecode, Message)
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MESSAGE_CLASS_DEFINITION(AMBEWorker::MsgTest, Message)
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AMBEWorker::AMBEWorker() :
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m_running(false),
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m_currentGainIn(0),
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m_currentGainOut(0),
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m_upsamplerLastValue(0.0f),
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m_phase(0),
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m_upsampling(1),
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m_volume(1.0f)
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{
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m_audioBuffer.resize(48000);
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m_audioBufferFill = 0;
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m_audioFifo = 0;
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std::fill(m_dvAudioSamples, m_dvAudioSamples+SerialDV::MBE_AUDIO_BLOCK_SIZE, 0);
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setVolumeFactors();
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}
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AMBEWorker::~AMBEWorker()
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{}
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bool AMBEWorker::open(const std::string& deviceRef)
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{
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return m_dvController.open(deviceRef);
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}
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void AMBEWorker::close()
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{
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m_dvController.close();
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}
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void AMBEWorker::process()
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{
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m_running = true;
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qDebug("AMBEWorker::process: started");
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while (m_running)
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{
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std::this_thread::sleep_for(std::chrono::seconds(1));
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}
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qDebug("AMBEWorker::process: stopped");
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emit finished();
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}
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void AMBEWorker::stop()
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{
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m_running = false;
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}
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void AMBEWorker::handleInputMessages()
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{
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Message* message;
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m_audioBufferFill = 0;
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AudioFifo *audioFifo = 0;
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while ((message = m_inputMessageQueue.pop()) != 0)
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{
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if (MsgMbeDecode::match(*message))
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{
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MsgMbeDecode *decodeMsg = (MsgMbeDecode *) message;
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int dBVolume = (decodeMsg->getVolumeIndex() - 30) / 4;
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float volume = pow(10.0, dBVolume / 10.0f);
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int upsampling = decodeMsg->getUpsampling();
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upsampling = upsampling > 6 ? 6 : upsampling < 1 ? 1 : upsampling;
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if ((volume != m_volume) || (upsampling != m_upsampling))
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{
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m_volume = volume;
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m_upsampling = upsampling;
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setVolumeFactors();
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}
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m_upsampleFilter.useHP(decodeMsg->getUseHP());
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if (m_dvController.decode(m_dvAudioSamples, decodeMsg->getMbeFrame(), decodeMsg->getMbeRate()))
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{
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if (upsampling > 1) {
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upsample(upsampling, m_dvAudioSamples, SerialDV::MBE_AUDIO_BLOCK_SIZE, decodeMsg->getChannels());
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} else {
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noUpsample(m_dvAudioSamples, SerialDV::MBE_AUDIO_BLOCK_SIZE, decodeMsg->getChannels());
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}
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audioFifo = decodeMsg->getAudioFifo();
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if (audioFifo && (m_audioBufferFill >= m_audioBuffer.size() - 960))
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{
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uint res = audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
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if (res != m_audioBufferFill) {
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qDebug("AMBEWorker::handleInputMessages: %u/%u audio samples written", res, m_audioBufferFill);
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}
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m_audioBufferFill = 0;
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}
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}
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else
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{
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qDebug("AMBEWorker::handleInputMessages: MsgMbeDecode: decode failed");
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}
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}
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delete message;
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if (m_inputMessageQueue.size() > 100)
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{
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qDebug("AMBEWorker::handleInputMessages: MsgMbeDecode: too many messages in queue. Flushing...");
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m_inputMessageQueue.clear();
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break;
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}
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}
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if (audioFifo)
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{
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uint res = audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
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if (res != m_audioBufferFill) {
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qDebug("AMBEWorker::handleInputMessages: %u/%u audio samples written", res, m_audioBufferFill);
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}
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m_audioBufferFill = 0;
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}
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m_timestamp = QDateTime::currentDateTime();
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}
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void AMBEWorker::pushMbeFrame(const unsigned char *mbeFrame,
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int mbeRateIndex,
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int mbeVolumeIndex,
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unsigned char channels,
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bool useHP,
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int upsampling,
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AudioFifo *audioFifo)
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{
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m_audioFifo = audioFifo;
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m_inputMessageQueue.push(MsgMbeDecode::create(mbeFrame, mbeRateIndex, mbeVolumeIndex, channels, useHP, upsampling, audioFifo));
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}
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bool AMBEWorker::isAvailable()
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{
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if (m_audioFifo == 0) {
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return true;
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}
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return m_timestamp.time().msecsTo(QDateTime::currentDateTime().time()) > 1000; // 1 second inactivity timeout
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}
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bool AMBEWorker::hasFifo(AudioFifo *audioFifo)
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{
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return m_audioFifo == audioFifo;
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}
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void AMBEWorker::upsample(int upsampling, short *in, int nbSamplesIn, unsigned char channels)
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{
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for (int i = 0; i < nbSamplesIn; i++)
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{
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//float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) m_compressor.compress(in[i])) : (float) m_compressor.compress(in[i]);
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float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
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float prev = m_upsamplerLastValue;
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qint16 upsample;
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for (int j = 1; j <= upsampling; j++)
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{
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upsample = (qint16) m_upsampleFilter.runLP(cur*m_upsamplingFactors[j] + prev*m_upsamplingFactors[upsampling-j]);
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m_audioBuffer[m_audioBufferFill].l = channels & 1 ? m_compressor.compress(upsample) : 0;
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m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? m_compressor.compress(upsample) : 0;
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if (m_audioBufferFill < m_audioBuffer.size() - 1) {
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++m_audioBufferFill;
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}
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}
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m_upsamplerLastValue = cur;
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}
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if (m_audioBufferFill >= m_audioBuffer.size() - 1) {
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qDebug("AMBEWorker::upsample(%d): audio buffer is full check its size", upsampling);
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}
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}
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void AMBEWorker::noUpsample(short *in, int nbSamplesIn, unsigned char channels)
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{
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for (int i = 0; i < nbSamplesIn; i++)
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{
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float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
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m_audioBuffer[m_audioBufferFill].l = channels & 1 ? cur*m_upsamplingFactors[0] : 0;
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m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? cur*m_upsamplingFactors[0] : 0;
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if (m_audioBufferFill < m_audioBuffer.size() - 1) {
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++m_audioBufferFill;
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}
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}
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if (m_audioBufferFill >= m_audioBuffer.size() - 1) {
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qDebug("AMBEWorker::noUpsample: audio buffer is full check its size");
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}
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}
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void AMBEWorker::setVolumeFactors()
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{
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m_upsamplingFactors[0] = m_volume;
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for (int i = 1; i <= m_upsampling; i++) {
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m_upsamplingFactors[i] = (i*m_volume) / (float) m_upsampling;
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}
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}
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