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Improve FEC operation with opus

merge-requests/9/head
Phil Taylor 2022-01-06 13:44:29 +00:00
rodzic 9284a49469
commit 6c29eb12a2
1 zmienionych plików z 32 dodań i 30 usunięć
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62
audiohandler.cpp
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@ -593,27 +593,29 @@ void audioHandler::incomingAudio(audioPacket inPacket)
return;
}
audioPacket livePacket = inPacket;
if (setup.codec == 0x40 || setup.codec == 0x80) {
unsigned char* in = (unsigned char*)inPacket.data.data();
/* Decode the frame. */
QByteArray outPacket((setup.samplerate / 50) * sizeof(qint16) * setup.radioChan, (char)0xff); // Preset the output buffer size.
qint16* out = (qint16*)outPacket.data();
int nSamples = opus_packet_get_nb_samples(in, inPacket.data.size(),setup.samplerate);
int nSamples = opus_packet_get_nb_samples(in, livePacket.data.size(),setup.samplerate);
if (nSamples != setup.samplerate / 50)
{
qInfo(logAudio()) << "Opus nSamples=" << nSamples << " expected:" << (setup.samplerate / 50);
return;
}
if (inPacket.seq > lastSentSeq + 1) {
nSamples = opus_decode(decoder, in, inPacket.data.size(), out, (setup.samplerate / 50), 1);
if (livePacket.seq > lastSentSeq + 1) {
nSamples = opus_decode(decoder, in, livePacket.data.size(), out, (setup.samplerate / 50), 1);
}
else {
nSamples = opus_decode(decoder, in, inPacket.data.size(), out, (setup.samplerate / 50), 0);
nSamples = opus_decode(decoder, in, livePacket.data.size(), out, (setup.samplerate / 50), 0);
}
if (nSamples < 0)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decode failed:" << opus_strerror(nSamples) << "packet size" << inPacket.data.length();
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decode failed:" << opus_strerror(nSamples) << "packet size" << livePacket.data.length();
return;
}
else {
@ -622,22 +624,22 @@ void audioHandler::incomingAudio(audioPacket inPacket)
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decoder mismatch: nBytes:" << nSamples * sizeof(qint16) * setup.radioChan << "outPacket:" << outPacket.size();
outPacket.resize(nSamples * sizeof(qint16) * setup.radioChan);
}
//qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decoded" << inPacket.data.size() << "bytes, into" << outPacket.length() << "bytes";
inPacket.data.clear();
inPacket.data = outPacket; // Replace incoming data with converted.
//qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decoded" << livePacket.data.size() << "bytes, into" << outPacket.length() << "bytes";
livePacket.data.clear();
livePacket.data = outPacket; // Replace incoming data with converted.
}
}
//qDebug(logAudio()) << "Got" << setup.bits << "bits, length" << inPacket.data.length();
//qDebug(logAudio()) << "Got" << setup.bits << "bits, length" << livePacket.data.length();
// Incoming data is 8bits?
if (setup.bits == 8)
{
// Current packet is 8bit so need to create a new buffer that is 16bit
QByteArray outPacket((int)inPacket.data.length() * 2 * (devChannels / setup.radioChan), (char)0xff);
QByteArray outPacket((int)livePacket.data.length() * 2 * (devChannels / setup.radioChan), (char)0xff);
qint16* out = (qint16*)outPacket.data();
for (int f = 0; f < inPacket.data.length(); f++)
for (int f = 0; f < livePacket.data.length(); f++)
{
int samp = (quint8)inPacket.data[f];
int samp = (quint8)livePacket.data[f];
for (int g = setup.radioChan; g <= devChannels; g++)
{
if (setup.ulaw)
@ -646,30 +648,30 @@ void audioHandler::incomingAudio(audioPacket inPacket)
*out++ = (qint16)((samp - 128) << 8) * this->volume;
}
}
inPacket.data.clear();
inPacket.data = outPacket; // Replace incoming data with converted.
livePacket.data.clear();
livePacket.data = outPacket; // Replace incoming data with converted.
}
else
{
// This is already a 16bit stream, do we need to convert to stereo?
if (setup.radioChan == 1 && devChannels > 1) {
// Yes
QByteArray outPacket(inPacket.data.length() * 2, (char)0xff); // Preset the output buffer size.
qint16* in = (qint16*)inPacket.data.data();
QByteArray outPacket(livePacket.data.length() * 2, (char)0xff); // Preset the output buffer size.
qint16* in = (qint16*)livePacket.data.data();
qint16* out = (qint16*)outPacket.data();
for (int f = 0; f < inPacket.data.length() / 2; f++)
for (int f = 0; f < livePacket.data.length() / 2; f++)
{
*out++ = (qint16)*in * this->volume;
*out++ = (qint16)*in++ * this->volume;
}
inPacket.data.clear();
inPacket.data = outPacket; // Replace incoming data with converted.
livePacket.data.clear();
livePacket.data = outPacket; // Replace incoming data with converted.
}
else
{
// We already have the same number of channels so just update volume.
qint16* in = (qint16*)inPacket.data.data();
for (int f = 0; f < inPacket.data.length() / 2; f++)
qint16* in = (qint16*)livePacket.data.data();
for (int f = 0; f < livePacket.data.length() / 2; f++)
{
*in = *in * this->volume;
in++;
@ -681,17 +683,17 @@ void audioHandler::incomingAudio(audioPacket inPacket)
/* We now have an array of 16bit samples in the NATIVE samplerate of the radio
If the radio sample rate is below 48000, we need to resample.
*/
//qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
//qDebug(logAudio()) << "Now 16 bit stereo, length" << livePacket.data.length();
if (resampleRatio != 1.0) {
// We need to resample
// We have a stereo 16bit stream.
quint32 outFrames = ((inPacket.data.length() / 2 / devChannels) * resampleRatio);
quint32 inFrames = (inPacket.data.length() / 2 / devChannels);
quint32 outFrames = ((livePacket.data.length() / 2 / devChannels) * resampleRatio);
quint32 inFrames = (livePacket.data.length() / 2 / devChannels);
QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.
const qint16* in = (qint16*)inPacket.data.constData();
const qint16* in = (qint16*)livePacket.data.constData();
qint16* out = (qint16*)outPacket.data();
int err = 0;
@ -699,17 +701,17 @@ void audioHandler::incomingAudio(audioPacket inPacket)
if (err) {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
inPacket.data.clear();
inPacket.data = outPacket; // Replace incoming data with converted.
livePacket.data.clear();
livePacket.data = outPacket; // Replace incoming data with converted.
}
//qDebug(logAudio()) << "Adding packet to buffer:" << inPacket.seq << ": " << inPacket.data.length();
//qDebug(logAudio()) << "Adding packet to buffer:" << livePacket.seq << ": " << livePacket.data.length();
if (!ringBuf->try_write(inPacket))
if (!ringBuf->try_write(livePacket))
{
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size();
}
if (inPacket.seq > lastSentSeq + 1) {
if ((inPacket.seq > lastSentSeq + 1) && (setup.codec == 0x40 || setup.codec == 0x80)) {
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Attempting FEC on packet" << inPacket.seq << "as last is"<<lastSentSeq ;
lastSentSeq = inPacket.seq;
incomingAudio(inPacket); // Call myself again to run the packet a second time (FEC)