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sdrangel/sdrbase/ambe/ambeworker.cpp

233 lines
7.7 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2019 F4EXB //
// written by Edouard Griffiths //
// //
// 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 as version 3 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 V3 for more details. //
// //
// You should have received a copy of the GNU General Public License //
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#include <algorithm>
#include <chrono>
#include <thread>
#include "audio/audiofifo.h"
#include "ambeworker.h"
MESSAGE_CLASS_DEFINITION(AMBEWorker::MsgMbeDecode, Message)
MESSAGE_CLASS_DEFINITION(AMBEWorker::MsgTest, Message)
AMBEWorker::AMBEWorker() :
m_running(false),
m_currentGainIn(0),
m_currentGainOut(0),
m_upsamplerLastValue(0.0f),
m_phase(0),
m_upsampling(1),
m_volume(1.0f)
{
m_audioBuffer.resize(48000);
m_audioBufferFill = 0;
m_audioFifo = 0;
std::fill(m_dvAudioSamples, m_dvAudioSamples+SerialDV::MBE_AUDIO_BLOCK_SIZE, 0);
setVolumeFactors();
}
AMBEWorker::~AMBEWorker()
{}
bool AMBEWorker::open(const std::string& deviceRef)
{
return m_dvController.open(deviceRef);
}
void AMBEWorker::close()
{
m_dvController.close();
}
void AMBEWorker::process()
{
m_running = true;
qDebug("AMBEWorker::process: started");
while (m_running)
{
std::this_thread::sleep_for(std::chrono::seconds(1));
}
qDebug("AMBEWorker::process: stopped");
emit finished();
}
void AMBEWorker::stop()
{
m_running = false;
}
void AMBEWorker::handleInputMessages()
{
Message* message;
m_audioBufferFill = 0;
AudioFifo *audioFifo = 0;
while ((message = m_inputMessageQueue.pop()) != 0)
{
if (MsgMbeDecode::match(*message))
{
MsgMbeDecode *decodeMsg = (MsgMbeDecode *) message;
int dBVolume = (decodeMsg->getVolumeIndex() - 30) / 4;
float volume = pow(10.0, dBVolume / 10.0f);
int upsampling = decodeMsg->getUpsampling();
upsampling = upsampling > 6 ? 6 : upsampling < 1 ? 1 : upsampling;
if ((volume != m_volume) || (upsampling != m_upsampling))
{
m_volume = volume;
m_upsampling = upsampling;
setVolumeFactors();
}
m_upsampleFilter.useHP(decodeMsg->getUseHP());
if (m_dvController.decode(m_dvAudioSamples, decodeMsg->getMbeFrame(), decodeMsg->getMbeRate()))
{
if (upsampling > 1) {
upsample(upsampling, m_dvAudioSamples, SerialDV::MBE_AUDIO_BLOCK_SIZE, decodeMsg->getChannels());
} else {
noUpsample(m_dvAudioSamples, SerialDV::MBE_AUDIO_BLOCK_SIZE, decodeMsg->getChannels());
}
audioFifo = decodeMsg->getAudioFifo();
if (audioFifo && (m_audioBufferFill >= m_audioBuffer.size() - 960))
{
uint res = audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
if (res != m_audioBufferFill) {
qDebug("AMBEWorker::handleInputMessages: %u/%u audio samples written", res, m_audioBufferFill);
}
m_audioBufferFill = 0;
}
}
else
{
qDebug("AMBEWorker::handleInputMessages: MsgMbeDecode: decode failed");
}
}
delete message;
if (m_inputMessageQueue.size() > 100)
{
qDebug("AMBEWorker::handleInputMessages: MsgMbeDecode: too many messages in queue. Flushing...");
m_inputMessageQueue.clear();
break;
}
}
if (audioFifo)
{
uint res = audioFifo->write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
if (res != m_audioBufferFill) {
qDebug("AMBEWorker::handleInputMessages: %u/%u audio samples written", res, m_audioBufferFill);
}
m_audioBufferFill = 0;
}
m_timestamp = QDateTime::currentDateTime();
}
void AMBEWorker::pushMbeFrame(const unsigned char *mbeFrame,
int mbeRateIndex,
int mbeVolumeIndex,
unsigned char channels,
bool useHP,
int upsampling,
AudioFifo *audioFifo)
{
m_audioFifo = audioFifo;
m_inputMessageQueue.push(MsgMbeDecode::create(mbeFrame, mbeRateIndex, mbeVolumeIndex, channels, useHP, upsampling, audioFifo));
}
bool AMBEWorker::isAvailable()
{
if (m_audioFifo == 0) {
return true;
}
return m_timestamp.time().msecsTo(QDateTime::currentDateTime().time()) > 1000; // 1 second inactivity timeout
}
bool AMBEWorker::hasFifo(AudioFifo *audioFifo)
{
return m_audioFifo == audioFifo;
}
void AMBEWorker::upsample(int upsampling, short *in, int nbSamplesIn, unsigned char channels)
{
for (int i = 0; i < nbSamplesIn; i++)
{
//float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) m_compressor.compress(in[i])) : (float) m_compressor.compress(in[i]);
float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
float prev = m_upsamplerLastValue;
qint16 upsample;
for (int j = 1; j <= upsampling; j++)
{
upsample = (qint16) m_upsampleFilter.runLP(cur*m_upsamplingFactors[j] + prev*m_upsamplingFactors[upsampling-j]);
m_audioBuffer[m_audioBufferFill].l = channels & 1 ? m_compressor.compress(upsample) : 0;
m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? m_compressor.compress(upsample) : 0;
if (m_audioBufferFill < m_audioBuffer.size() - 1) {
++m_audioBufferFill;
}
}
m_upsamplerLastValue = cur;
}
if (m_audioBufferFill >= m_audioBuffer.size() - 1) {
qDebug("AMBEWorker::upsample(%d): audio buffer is full check its size", upsampling);
}
}
void AMBEWorker::noUpsample(short *in, int nbSamplesIn, unsigned char channels)
{
for (int i = 0; i < nbSamplesIn; i++)
{
float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
m_audioBuffer[m_audioBufferFill].l = channels & 1 ? cur*m_upsamplingFactors[0] : 0;
m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? cur*m_upsamplingFactors[0] : 0;
if (m_audioBufferFill < m_audioBuffer.size() - 1) {
++m_audioBufferFill;
}
}
if (m_audioBufferFill >= m_audioBuffer.size() - 1) {
qDebug("AMBEWorker::noUpsample: audio buffer is full check its size");
}
}
void AMBEWorker::setVolumeFactors()
{
m_upsamplingFactors[0] = m_volume;
for (int i = 1; i <= m_upsampling; i++) {
m_upsamplingFactors[i] = (i*m_volume) / (float) m_upsampling;
}
}