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Use preferred sample rate rather than force 48000

merge-requests/5/head
Phil Taylor 2021-05-27 23:23:23 +01:00
rodzic 5e530b2eba
commit d072fdc4f3
1 zmienionych plików z 23 dodań i 19 usunięć
Pokaż statystyki Ściągnij plik aktualizacji Ściągnij plik porównania Expand all files Collapse all files

42
audiohandler.cpp
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@ -84,6 +84,10 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
(info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") <<
(info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : "");
qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate;
if (isInput)
qInfo(logAudio()) << " Input Channels:" << info.inputChannels;
else
qInfo(logAudio()) << " Output Channels:" << info.outputChannels;
}
qInfo(logAudio()) << " chunkSize: " << chunkSize;
@ -96,10 +100,10 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
int resample_error = 0;
if (isInput) {
resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error);
resampler = wf_resampler_init(radioChannels, info.preferredSampleRate, samplerate, resampleQuality, &resample_error);
try {
aParams.nChannels = 1; // Internally this is always 2 channels
audio.openStream(NULL, &aParams, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &staticWrite, this);
aParams.nChannels = radioChannels;
audio.openStream(NULL, &aParams, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticWrite, this);
audio.startStream();
}
catch (RtAudioError& e) {
@ -109,11 +113,11 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
}
else
{
resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error);
resampler = wf_resampler_init(radioChannels, samplerate, info.preferredSampleRate, resampleQuality, &resample_error);
try {
unsigned int length = chunkSize / 2;
aParams.nChannels = 2; // Internally this is always 2 channels
audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &length, &staticRead, this);
audio.openStream(&aParams, NULL, RTAUDIO_SINT16, info.preferredSampleRate, &length, &staticRead, this);
audio.startStream();
}
catch (RtAudioError& e) {
@ -176,16 +180,16 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
memcpy(buffer + sentlen, tempBuf.data.constData() + tempBuf.sent, send);
tempBuf.sent = tempBuf.sent + send;
sentlen = sentlen + send;
qDebug(logAudio()) << "Adding partial:" << send;
//qDebug(logAudio()) << "Adding partial:" << send;
}
if (currentLatency > (int)audioLatency) {
//qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq <<
// " arrived too late (increase output latency!) " <<
// dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
//lastSeq = packet.seq;
//if (!ringBuf->try_read(packet))
// return sentlen;
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq <<
" arrived too late (increase output latency!) " <<
dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
lastSeq = packet.seq;
if (!ringBuf->try_read(packet))
return sentlen;
currentLatency = packet.time.msecsTo(QTime::currentTime());
}
@ -194,18 +198,18 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
sentlen = sentlen + send;
if (send < packet.data.length())
{
qDebug(logAudio()) << "Asking for partial, sent:" << send << "packet length" << packet.data.length();
//qDebug(logAudio()) << "Asking for partial, sent:" << send << "packet length" << packet.data.length();
tempBuf = packet;
tempBuf.sent = tempBuf.sent + send;
//lastSeq = packet.seq;
//break;
lastSeq = packet.seq;
break;
}
if (packet.seq <= lastSeq) {
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Duplicate/early audio packet: " << hex << lastSeq << " got " << hex << packet.seq;
qDebug(logAudio()) << (isInput ? "Input" : "Output") << "Duplicate/early audio packet: " << hex << lastSeq << " got " << hex << packet.seq;
}
else if (packet.seq != lastSeq + 1) {
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet.seq - 1;
qDebug(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet.seq - 1;
}
lastSeq = packet.seq;
}
@ -219,7 +223,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
{
int sentlen = 0;
unsigned int nBytes = nFrames * 2; // This is ALWAYS 2 bytes per sample and 1 channel
unsigned int nBytes = nFrames * 2 * radioChannels; // This is ALWAYS 2 bytes per sample
const char* data = (const char*)inputBuffer;
while (sentlen < nBytes) {
if (tempBuf.sent != chunkSize)
@ -229,7 +233,7 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int
sentlen = sentlen + send;
tempBuf.seq = 0; // Not used in TX
tempBuf.time = QTime::currentTime();
tempBuf.sent = tempBuf.data.length();
tempBuf.sent = tempBuf.sent + send;
}
else {
if (!ringBuf->try_write(tempBuf))