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Lots more changes for rtaudio compatibility

merge-requests/5/head
Phil Taylor 2021-05-29 18:59:45 +01:00
rodzic 7eeb5f08db
commit 923adbaa30
2 zmienionych plików z 135 dodań i 78 usunięć
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211
audiohandler.cpp
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@ -32,7 +32,7 @@ audioHandler::~audioHandler()
delete ringBuf;
}
bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, int port, quint8 resampleQuality)
bool audioHandler::init(const quint8 bits, const quint8 radioChan, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, int port, quint8 resampleQuality)
{
if (isInitialized) {
return false;
@ -43,10 +43,9 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
this->isInput = isinput;
this->radioSampleBits = bits;
this->radioSampleRate = samplerate;
this->radioChannels = channels;
this->radioChannels = radioChan;
// chunk size is always relative to Internal Sample Rate.
this->chunkSize = (INTERNAL_SAMPLE_RATE / 25) * radioChannels;
ringBuf = new wilt::Ring<audioPacket>(100); // Should be customizable.
tempBuf.sent = 0;
@ -61,6 +60,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
aParams.deviceId = audio.getDefaultOutputDevice();
}
aParams.firstChannel = 0;
aParams.nChannels = 2; // Internally this is always 2 channels for TX and RX.
try {
info = audio.getDeviceInfo(aParams.deviceId);
@ -72,6 +72,9 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
if (info.probed)
{
// Per channel chunk size.
this->chunkSize = (info.preferredSampleRate / 50);
qInfo(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed";
if (info.nativeFormats == 0)
qInfo(logAudio()) << " No natively supported data formats!";
@ -100,9 +103,8 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
int resample_error = 0;
if (isInput) {
resampler = wf_resampler_init(radioChannels, info.preferredSampleRate, samplerate, resampleQuality, &resample_error);
resampler = wf_resampler_init(2, info.preferredSampleRate, samplerate, resampleQuality, &resample_error);
try {
aParams.nChannels = radioChannels;
audio.openStream(NULL, &aParams, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticWrite, this);
audio.startStream();
}
@ -113,11 +115,9 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
}
else
{
resampler = wf_resampler_init(radioChannels, samplerate, info.preferredSampleRate, resampleQuality, &resample_error);
resampler = wf_resampler_init(2, 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, info.preferredSampleRate, &length, &staticRead, this);
audio.openStream(&aParams, NULL, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticRead, this);
audio.startStream();
}
catch (RtAudioError& e) {
@ -127,6 +127,8 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
}
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "device successfully opened";
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "detected latency:" <<audio.getStreamLatency();
wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen;
@ -135,8 +137,8 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
void audioHandler::setVolume(unsigned char volume)
{
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume / 255.0) << ")";
this->volume = (qreal)(volume/255.0);
this->volume = (qreal)volume/255.0;
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << this->volume << ")";
}
@ -149,26 +151,27 @@ void audioHandler::setVolume(unsigned char volume)
/// <returns></returns>
int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
{
Q_UNUSED(inputBuffer);
Q_UNUSED(streamTime);
// Calculate output length, always full samples
int sentlen = 0;
quint8* buffer = (quint8*)outputBuffer;
if (status == RTAUDIO_OUTPUT_UNDERFLOW)
qDebug(logAudio()) << "Underflow detected";
unsigned int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample and 2 channels
int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample and 2 channels
if (ringBuf->size()>0)
{
// Output buffer is ALWAYS 16 bit.
//qDebug(logAudio()) << "Read: nFrames" << nFrames << "nBytes" << nBytes;
while (sentlen < nBytes)
{
audioPacket packet;
if (!ringBuf->try_read(packet))
{
qDebug() << "No more data available but buffer is not full! sentlen:" << sentlen << " nBytes:" << nBytes ;
return 0;
break;
}
currentLatency = packet.time.msecsTo(QTime::currentTime());
@ -183,7 +186,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
if (tempBuf.sent != tempBuf.data.length())
{
// We still don't have enough buffer space for this?
return 0;
break;
}
//qDebug(logAudio()) << "Adding partial:" << send;
}
@ -194,7 +197,7 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
lastSeq = packet.seq;
if (!ringBuf->try_read(packet))
return sentlen;
break;
currentLatency = packet.time.msecsTo(QTime::currentTime());
}
@ -227,13 +230,16 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
{
Q_UNUSED(outputBuffer);
Q_UNUSED(streamTime);
int sentlen = 0;
unsigned int nBytes = nFrames * 2 * radioChannels; // This is ALWAYS 2 bytes per sample
int nBytes = nFrames * 2 * 2; // This is ALWAYS 2 bytes per sample
const char* data = (const char*)inputBuffer;
//qDebug(logAudio()) << "nFrames" << nFrames << "nBytes" << nBytes;
while (sentlen < nBytes) {
if (tempBuf.sent != chunkSize)
if (tempBuf.sent != nBytes)
{
int send = qMin((int)(nBytes - sentlen), (int)chunkSize - tempBuf.sent);
int send = qMin((int)(nBytes - sentlen), (int)nBytes - tempBuf.sent);
tempBuf.data.append(QByteArray::fromRawData(data + sentlen, send));
sentlen = sentlen + send;
tempBuf.seq = 0; // Not used in TX
@ -251,6 +257,8 @@ int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int
}
}
//qDebug(logAudio()) << "sentlen" << sentlen;
return 0;
}
@ -276,85 +284,113 @@ void audioHandler::stateChanged(QAudio::State state)
void audioHandler::incomingAudio(audioPacket data)
void audioHandler::incomingAudio(audioPacket inPacket)
{
// No point buffering audio until stream is actually running.
// Regardless of the radio stream format, the buffered audio will ALWAYS be
// 16bit sample interleaved stereo 48K (or whatever the native sample rate is)
if (!audio.isStreamRunning())
{
qDebug(logAudio()) << "Packet received before stream was started";
return;
}
//qDebug(logAudio()) << "Got" << radioSampleBits << "bits, length" << inPacket.data.length();
// Incoming data is 8bits?
if (radioSampleBits == 8)
{
QByteArray outPacket((int)data.data.length() * 2, (char)0xff);
// Current packet is 8bit so need to create a new buffer that is 16bit
QByteArray outPacket((int)inPacket.data.length() * 2 *(2/radioChannels), (char)0xff);
qint16* out = (qint16*)outPacket.data();
for (int f = 0; f < data.data.length(); f++)
for (int f = 0; f < inPacket.data.length(); f++)
{
if (isUlaw)
{
out[f] = ulaw_decode[(quint8)data.data[f]];
if (radioChannels == 1)
{
*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
}
else
{
// This is already 2 channel.
*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
}
}
else
{
// Convert 8-bit sample to 16-bit
out[f] = (qint16)(((quint8)data.data[f] << 8) - 32640);
if (radioChannels == 1)
{
*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640) * this->volume;
*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640) * this->volume;
}
else
{
*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640 * this->volume);
}
}
}
data.data.clear();
data.data = outPacket; // Replace incoming data with converted.
inPacket.data.clear();
inPacket.data = outPacket; // Replace incoming data with converted.
}
else
{
// This is already a 16bit stream, do we need to convert to stereo?
if (radioChannels == 1) {
// Yes
QByteArray outPacket(inPacket.data.length() * 2, (char)0xff); // Preset the output buffer size.
qint16* in = (qint16*)inPacket.data.data();
qint16* out = (qint16*)outPacket.data();
for (int f = 0; f < inPacket.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.
}
else
{
// We already have two channels so just update volume.
qint16* in = (qint16*)inPacket.data.data();
for (int f = 0; f < inPacket.data.length() / 2; f++)
{
*in = *in++ * this->volume;
}
}
//qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.data.length();
}
/* 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();
if (ratioDen != 1) {
// We need to resample
quint32 outFrames = ((data.data.length() / 2) * ratioDen) / radioChannels;
quint32 inFrames = (data.data.length() / 2) / radioChannels;
QByteArray outPacket(outFrames * 2 * radioChannels, 0xff); // Preset the output buffer size.
// We have a stereo 16bit stream.
quint32 outFrames = ((inPacket.data.length() / 4) * ratioDen);
quint32 inFrames = (inPacket.data.length() / 4);
QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.
const qint16* in = (qint16*)inPacket.data.constData();
qint16* out = (qint16*)outPacket.data();
int err = 0;
if (this->radioChannels == 1) {
err = wf_resampler_process_int(resampler, 0, (const qint16*)data.data.constData(), &inFrames, (qint16*)outPacket.data(), &outFrames);
}
else {
err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.data.constData(), &inFrames, (qint16*)outPacket.data(), &outFrames);
}
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
if (err) {
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
data.data.clear();
data.data = outPacket; // Replace incoming data with converted.
inPacket.data.clear();
inPacket.data = outPacket; // Replace incoming data with converted.
}
if (radioChannels == 1) {
// Convert to stereo and set volume.
QByteArray outPacket(data.data.length()*2, 0xff); // Preset the output buffer size.
qint16* in = (qint16*)data.data.data();
qint16* out = (qint16*)outPacket.data();
for (int f = 0; f < data.data.length()/2; f++)
{
*out++ = *in * volume;
*out++ = *in++ * volume;
}
data.data.clear();
data.data = outPacket; // Replace incoming data with converted.
} else {
// We already have two channels so just update volume.
qint16* in = (qint16*)data.data.data();
for (int f = 0; f < data.data.length() / 2; f++)
{
in[f] = in[f] * volume;
}
}
//qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.data.length();
//qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
if (!ringBuf->try_write(data))
if (!ringBuf->try_write(inPacket))
{
qDebug(logAudio()) << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size();
}
@ -381,38 +417,57 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
{
audioPacket packet;
packet.sent = 0;
if (ringBuf != Q_NULLPTR && ringBuf->try_read(packet))
{
//qDebug(logAudio) << "Chunksize" << this->chunkSize << "Packet size" << packet.data.length();
// Packet will arrive as stereo interleaved 16bit 48K
if (ratioNum != 1)
{
// We need to resample (we are STILL 16 bit!)
quint32 outFrames = ((packet.data.length() / 2) / ratioNum) / radioChannels;
quint32 inFrames = (packet.data.length() / 2) / radioChannels;
QByteArray inPacket((int)outFrames * 2 * radioChannels, (char)0xff);
quint32 outFrames = ((packet.data.length() / 4) / ratioNum);
quint32 inFrames = (packet.data.length() / 4);
QByteArray outPacket((int)outFrames * 4, (char)0xff);
const qint16* in = (qint16*)packet.data.constData();
qint16* out = (qint16*)inPacket.data();
qint16* out = (qint16*)outPacket.data();
int err = 0;
if (this->radioChannels == 1) {
err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames);
}
else {
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
}
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
if (err) {
qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
//qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
//qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length();
packet.data.clear();
packet.data = inPacket; // Copy output packet back to input buffer.
packet.data = outPacket; // Copy output packet back to input buffer.
}
//qDebug(logAudio()) << "Now resampled, length" << packet.data.length();
// Do we need to convert mono to stereo?
if (radioChannels == 1)
{
// Strip out right channel?
QByteArray outPacket(packet.data.length()/2, (char)0xff);
const qint16* in = (qint16*)packet.data.constData();
qint16* out = (qint16*)outPacket.data();
for (int f = 0; f < outPacket.length()/2; f++)
{
*out++ = *in++;
*++in; // Skip each even channel.
}
packet.data.clear();
packet.data = outPacket; // Copy output packet back to input buffer.
}
//qDebug(logAudio()) << "Now mono, length" << packet.data.length();
// Do we need to convert 16-bit to 8-bit?
if (radioSampleBits == 8) {
QByteArray inPacket((int)packet.data.length() / 2, (char)0xff);
QByteArray outPacket((int)packet.data.length() / 2, (char)0xff);
qint16* in = (qint16*)packet.data.data();
for (int f = 0; f < inPacket.length(); f++)
for (int f = 0; f < outPacket.length(); f++)
{
quint8 outdata = 0;
if (isUlaw) {
@ -425,14 +480,16 @@ void audioHandler::getNextAudioChunk(QByteArray& ret)
else {
outdata = (quint8)(((qFromLittleEndian<qint16>(in + f) >> 8) ^ 0x80) & 0xff);
}
inPacket[f] = (char)outdata;
outPacket[f] = (char)outdata;
}
packet.data.clear();
packet.data = inPacket; // Copy output packet back to input buffer.
packet.data = outPacket; // Copy output packet back to input buffer.
}
ret = packet.data;
//qDebug(logAudio()) << "Now radio format, length" << packet.data.length();
}
return;
}

2
audiohandler.h
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@ -7,6 +7,7 @@
#include <QByteArray>
#include <QMutex>
#include <QtEndian>
#include <QtMath>
#include "rtaudio/RtAudio.h"
typedef signed short MY_TYPE;
@ -119,7 +120,6 @@ private:
audioPacket tempBuf;
quint16 currentLatency;
qreal volume=1.0;
};
#endif // AUDIOHANDLER_H