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Change TX audio to use timed buffer.

merge-requests/2/head
Phil Taylor 2021-02-27 09:34:56 +00:00
rodzic 5a547440b3
commit e593e5e90a
4 zmienionych plików z 82 dodań i 61 usunięć
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131
audiohandler.cpp
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@ -844,7 +844,6 @@ void audioHandler::reinit()
delete audioOutput;
audioOutput = Q_NULLPTR;
audioOutput = new QAudioOutput(deviceInfo, format, this);
//audioOutput->setLatency(audioBuffer);
connect(audioOutput, SIGNAL(notify()), SLOT(notified()));
connect(audioOutput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
}
@ -869,7 +868,8 @@ void audioHandler::start()
else {
this->open(QIODevice::ReadOnly | QIODevice::Unbuffered);
audioOutput->start(this);
}
audioOutput->suspend(); // Suspend until data has arrived to "kick-start" playback
}
}
void audioHandler::setVolume(float volume)
@ -888,26 +888,23 @@ void audioHandler::flush()
else {
audioOutput->reset();
}
buffer.clear();
QMutexLocker locker(&mutex);
audioBuffer.clear();
this->start();
}
void audioHandler::stop()
{
//QMutexLocker locker(&mutex);
if (audioOutput && audioOutput->state() != QAudio::StoppedState) {
// Stop audio output
audioOutput->stop();
QByteArray ret;
audioBuffer.clear();
//buffer.clear();
this->close();
}
if (audioInput && audioInput->state() != QAudio::StoppedState) {
// Stop audio output
audioInput->stop();
buffer.clear();
this->close();
}
}
@ -921,7 +918,10 @@ qint64 audioHandler::readData(char* data, qint64 maxlen)
if (!audioBuffer.isEmpty())
{
// Sort the buffer by seq number.
// We must lock the mutex for the entire time that the buffer may be modified.
QMutexLocker locker(&mutex);
// Sort the buffer by seq number. This is important and audio packets may have arrived out-of-order
std::sort(audioBuffer.begin(), audioBuffer.end(),
[](const AUDIOPACKET& a, const AUDIOPACKET& b) -> bool
{
@ -932,12 +932,11 @@ qint64 audioHandler::readData(char* data, qint64 maxlen)
int divisor = 16 / radioSampleBits;
auto packet = audioBuffer.begin();
while (packet != audioBuffer.end() && sentlen<maxlen)
while (packet != audioBuffer.end() && sentlen < maxlen)
{
if (packet->time.msecsTo(QTime::currentTime()) > latency) {
//qDebug(logAudio()) << "Packet " << hex << packet->seq << " arrived too late (increase rx buffer size!) " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms";
QMutexLocker locker(&mutex);
packet=audioBuffer.erase(packet); // returns next packet
packet = audioBuffer.erase(packet); // returns next packet
}
else
{
@ -971,7 +970,6 @@ qint64 audioHandler::readData(char* data, qint64 maxlen)
if (send == packet->data.length())
{
QMutexLocker locker(&mutex);
packet = audioBuffer.erase(packet); // returns next packet
if (maxlen - sentlen == 0)
{
@ -992,7 +990,6 @@ qint64 audioHandler::readData(char* data, qint64 maxlen)
}
}
}
//qDebug(logAudio()) << "Returning: " << sentlen << " max: " << maxlen;
return sentlen;
}
@ -1000,39 +997,58 @@ qint64 audioHandler::readData(char* data, qint64 maxlen)
qint64 audioHandler::writeData(const char* data, qint64 len)
{
QMutexLocker locker(&mutex);
if (isUlaw) {
for (int f = 0; f < len / 2; f++)
{
qint16 enc = qFromLittleEndian<quint16>(data + f * 2);
if (enc >=0)
buffer.append((quint8)ulaw_encode[enc]);
else
buffer.append((quint8) 0x7f & ulaw_encode[-enc]);
// The radio expects packets in radioSampleBits * 120 size packets.
int chunkSize = radioSampleBits * 120;
int multiplier = 16 / radioSampleBits;
int sentlen = 0;
while (sentlen < len) {
QByteArray buffer;
if (radioSampleBits == 8) {
int f = 0;
while (f < chunkSize && f * multiplier + sentlen < len)
{
if (isUlaw) {
qint16 enc = qFromLittleEndian<quint16>(data + (f * multiplier + sentlen));
if (enc >= 0)
buffer.append((quint8)ulaw_encode[enc]);
else
buffer.append((quint8)0x7f & ulaw_encode[-enc]);
}
else {
buffer.append((quint8)(((qFromLittleEndian<qint16>(data + (f * multiplier + sentlen)) >> 8) ^ 0x80) & 0xff));
}
f++;
}
}
}
else if (radioSampleBits == 8) {
for (int f = 0; f < len / 2; f++)
{
buffer.append((quint8)(((qFromLittleEndian<qint16>(data + f * 2) >> 8) ^ 0x80) & 0xff));
}
}
else if (radioSampleBits == 16) {
buffer.append(QByteArray::fromRawData(data, len));
}
else {
qWarning() << "Unsupported number of bits! :" << radioSampleBits;
}
if (buffer.size() >= radioSampleBits * 120) {
chunkAvailable = true;
}
return (len); // Always return the same number as we received
else if (radioSampleBits == 16)
{
buffer.append(QByteArray::fromRawData(data+sentlen, chunkSize*multiplier));
}
sentlen = sentlen + (chunkSize*multiplier);
AUDIOPACKET tempAudio;
tempAudio.seq = 0; // Not used in TX
tempAudio.time = QTime::currentTime();
tempAudio.sent = 0;
tempAudio.data = buffer;
QMutexLocker locker(&mutex);
audioBuffer.append(tempAudio);
if (buffer.length() < chunkSize)
{
// We don't have enough for a full packet yet, what to do?
qDebug(logAudio) << "*** Small Packet *** " << buffer.length() << " less than " << chunkSize;
}
}
return (sentlen); // Always return the same number as we received
}
qint64 audioHandler::bytesAvailable() const
{
return buffer.length() + QIODevice::bytesAvailable();
return 0;
}
bool audioHandler::isSequential() const
@ -1049,10 +1065,7 @@ void audioHandler::stateChanged(QAudio::State state)
{
if (state == QAudio::IdleState && audioOutput->error() == QAudio::UnderrunError) {
qDebug(logAudio()) << this->metaObject()->className() << "RX:Buffer underrun";
#ifdef Q_OS_WIN
QMutexLocker locker(&mutex);
audioOutput->suspend();
#endif
}
//qDebug(logAudio()) << this->metaObject()->className() << ": state = " << state;
}
@ -1064,6 +1077,7 @@ void audioHandler::incomingAudio(const AUDIOPACKET data)
if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) {
QMutexLocker locker(&mutex);
audioBuffer.push_back(data);
// Restart playback
if (audioOutput->state() == QAudio::SuspendedState)
{
@ -1086,21 +1100,28 @@ void audioHandler::getLatency()
bool audioHandler::isChunkAvailable()
{
return chunkAvailable;
return (!audioBuffer.isEmpty());
}
void audioHandler::getNextAudioChunk(QByteArray& ret)
{
quint16 numSamples = radioSampleBits * 120;
if (buffer.size() >= numSamples) {
ret.append(buffer.mid(0, numSamples));
QMutexLocker locker(&mutex);
buffer.remove(0, numSamples);
}
if (buffer.size() < numSamples)
{
chunkAvailable = false;
}
if (!audioBuffer.isEmpty())
{
QMutexLocker locker(&mutex);
auto packet = audioBuffer.begin();
while (packet != audioBuffer.end())
{
if (packet->time.msecsTo(QTime::currentTime()) > 40) {
qDebug(logAudio()) << "TX Packet arrived too late " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms";
packet = audioBuffer.erase(packet); // returns next packet
}
else {
ret.append(packet->data);
packet = audioBuffer.erase(packet); // returns next packet
break; // We only want one packet.
}
}
}
return;
}

1
audiohandler.h
Wyświetl plik

@ -82,7 +82,6 @@ private:
bool isInput; // Used to determine whether input or output audio
float volume;
QByteArray buffer;
QAudioFormat format;
QAudioDeviceInfo deviceInfo;
quint16 radioSampleRate;

5
udphandler.cpp
Wyświetl plik

@ -452,7 +452,7 @@ void udpHandler::sendToken(uint8_t magic)
authInnerSendSeq++;
sendTrackedPacket(QByteArray::fromRawData((const char *)p.packet, sizeof(p)));
// The radio should request a repeat of the token renewal!
// The radio should request a repeat of the token renewal packet via retransmission!
//tokenTimer->start(100); // Set 100ms timer for retry (this will be cancelled if a response is received)
return;
}
@ -839,9 +839,10 @@ void udpAudio::dataReceived()
tempAudio.time = lastReceived;
tempAudio.sent = 0;
tempAudio.data = r.mid(24);
// Prefer signal/slot to forward audio as it is thread/safe
// Need to do more testing but latency appears fine.
emit haveAudioData(tempAudio);
//rxaudio->incomingAudio(tempAudio);
//rxaudio->incomingAudio(r.mid(24));
}
}
break;

6
wfmain.cpp
Wyświetl plik

@ -776,9 +776,9 @@ void wfmain::loadSettings()
ui->rxLatencySlider->setTracking(false); // Stop it sending value on every change.
prefs.audioTXLatency = settings.value("AudioTXLatency", defPrefs.audioTXLatency).toInt();
ui->rxLatencySlider->setEnabled(ui->lanEnableChk->isChecked());
ui->rxLatencySlider->setValue(prefs.audioTXLatency);
ui->rxLatencySlider->setTracking(false); // Stop it sending value on every change.
ui->txLatencySlider->setEnabled(ui->lanEnableChk->isChecked());
ui->txLatencySlider->setValue(prefs.audioTXLatency);
ui->txLatencySlider->setTracking(false); // Stop it sending value on every change.
prefs.audioRXSampleRate = settings.value("AudioRXSampleRate", defPrefs.audioRXSampleRate).toInt();
prefs.audioTXSampleRate = settings.value("AudioTXSampleRate", defPrefs.audioTXSampleRate).toInt();