wfview/audiohandler.cpp

926 wiersze
28 KiB
C++

/*
This class handles both RX and TX audio, each is created as a seperate instance of the class
but as the setup/handling if output (RX) and input (TX) devices is so similar I have combined them.
*/
#include "audiohandler.h"
#include "logcategories.h"
#include "ulaw.h"
#if defined(Q_OS_WIN) && defined(PORTAUDIO)
#include <objbase.h>
#endif
audioHandler::audioHandler(QObject* parent)
{
Q_UNUSED(parent)
}
audioHandler::~audioHandler()
{
if (isInitialized) {
#if defined(RTAUDIO)
try {
audio->abortStream();
audio->closeStream();
}
catch (RtAudioError& e) {
qInfo(logAudio()) << "Error closing stream:" << aParams.deviceId << ":" << QString::fromStdString(e.getMessage());
}
delete audio;
#elif defined(PORTAUDIO)
Pa_StopStream(audio);
Pa_CloseStream(audio);
#else
stop();
#endif
}
if (ringBuf != Q_NULLPTR) {
delete ringBuf;
}
if (resampler != Q_NULLPTR) {
speex_resampler_destroy(resampler);
qDebug(logAudio()) << "Resampler closed";
}
if (encoder != Q_NULLPTR) {
qInfo(logAudio()) << "Destroying opus encoder";
opus_encoder_destroy(encoder);
}
if (decoder != Q_NULLPTR) {
qInfo(logAudio()) << "Destroying opus decoder";
opus_decoder_destroy(decoder);
}
}
bool audioHandler::init(audioSetup setupIn)
{
if (isInitialized) {
return false;
}
/*
0x01 uLaw 1ch 8bit
0x02 PCM 1ch 8bit
0x04 PCM 1ch 16bit
0x08 PCM 2ch 8bit
0x10 PCM 2ch 16bit
0x20 uLaw 2ch 8bit
*/
setup = setupIn;
setup.radioChan = 1;
setup.bits = 8;
if (setup.codec == 0x01 || setup.codec == 0x20) {
setup.ulaw = true;
}
if (setup.codec == 0x08 || setup.codec == 0x10 || setup.codec == 0x20 || setup.codec == 0x80) {
setup.radioChan = 2;
}
if (setup.codec == 0x04 || setup.codec == 0x10 || setup.codec == 0x40 || setup.codec == 0x80) {
setup.bits = 16;
}
ringBuf = new wilt::Ring<audioPacket>(setupIn.latency / 8 + 1); // Should be customizable.
tempBuf.sent = 0;
if(!setup.isinput)
{
this->setVolume(setup.localAFgain);
}
#if defined(RTAUDIO)
#if !defined(Q_OS_MACX)
options.flags = ((!RTAUDIO_HOG_DEVICE) | (RTAUDIO_MINIMIZE_LATENCY));
#endif
#if defined(Q_OS_LINUX)
audio = new RtAudio(RtAudio::Api::LINUX_ALSA);
#elif defined(Q_OS_WIN)
audio = new RtAudio(RtAudio::Api::WINDOWS_WASAPI);
#elif defined(Q_OS_MACX)
audio = new RtAudio(RtAudio::Api::MACOSX_CORE);
#endif
if (setup.port > 0) {
aParams.deviceId = setup.port;
}
else if (setup.isinput) {
aParams.deviceId = audio->getDefaultInputDevice();
}
else {
aParams.deviceId = audio->getDefaultOutputDevice();
}
aParams.firstChannel = 0;
try {
info = audio->getDeviceInfo(aParams.deviceId);
}
catch (RtAudioError& e) {
qInfo(logAudio()) << "Device error:" << aParams.deviceId << ":" << QString::fromStdString(e.getMessage());
return isInitialized;
}
if (info.probed)
{
// Always use the "preferred" sample rate
// We can always resample if needed
this->nativeSampleRate = info.preferredSampleRate;
// Per channel chunk size.
this->chunkSize = (this->nativeSampleRate / 50);
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed";
if (info.nativeFormats == 0)
{
qInfo(logAudio()) << " No natively supported data formats!";
return false;
}
else {
qDebug(logAudio()) << " Supported formats:" <<
(info.nativeFormats & RTAUDIO_SINT8 ? "8-bit int," : "") <<
(info.nativeFormats & RTAUDIO_SINT16 ? "16-bit int," : "") <<
(info.nativeFormats & RTAUDIO_SINT24 ? "24-bit int," : "") <<
(info.nativeFormats & RTAUDIO_SINT32 ? "32-bit int," : "") <<
(info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") <<
(info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : "");
qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate;
if (setup.isinput) {
devChannels = info.inputChannels;
}
else {
devChannels = info.outputChannels;
}
qInfo(logAudio()) << " Channels:" << devChannels;
if (devChannels > 2) {
devChannels = 2;
}
aParams.nChannels = devChannels;
}
qInfo(logAudio()) << " chunkSize: " << chunkSize;
try {
if (setup.isinput) {
audio->openStream(NULL, &aParams, RTAUDIO_SINT16, this->nativeSampleRate, &this->chunkSize, &staticWrite, this, &options);
}
else {
audio->openStream(&aParams, NULL, RTAUDIO_SINT16, this->nativeSampleRate, &this->chunkSize, &staticRead, this, &options);
}
audio->startStream();
isInitialized = true;
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "device successfully opened";
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "detected latency:" << audio->getStreamLatency();
}
catch (RtAudioError& e) {
qInfo(logAudio()) << "Error opening:" << QString::fromStdString(e.getMessage());
}
}
else
{
qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") could not be probed, check audio configuration!";
}
#elif defined(PORTAUDIO)
PaError err;
#ifdef Q_OS_WIN
CoInitialize(0);
#endif
memset(&aParams, 0,sizeof(PaStreamParameters));
if (setup.port > 0) {
aParams.device = setup.port;
}
else if (setup.isinput) {
aParams.device = Pa_GetDefaultInputDevice();
}
else {
aParams.device = Pa_GetDefaultOutputDevice();
}
info = Pa_GetDeviceInfo(aParams.device);
aParams.channelCount = 2;
aParams.hostApiSpecificStreamInfo = NULL;
aParams.sampleFormat = paInt16;
if (setup.isinput) {
aParams.suggestedLatency = info->defaultLowInputLatency;
}
else {
aParams.suggestedLatency = info->defaultLowOutputLatency;
}
aParams.hostApiSpecificStreamInfo = NULL;
// Always use the "preferred" sample rate (unless it is 44100)
// We can always resample if needed
if (info->defaultSampleRate == 44100) {
this->nativeSampleRate = 48000;
}
else {
this->nativeSampleRate = info->defaultSampleRate;
}
// Per channel chunk size.
this->chunkSize = (this->nativeSampleRate / 50);
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << info->name << "(" << aParams.device << ") successfully probed";
if (setup.isinput) {
devChannels = info->maxInputChannels;
}
else {
devChannels = info->maxOutputChannels;
}
if (devChannels > 2) {
devChannels = 2;
}
aParams.channelCount = devChannels;
qInfo(logAudio()) << " Channels:" << devChannels;
qInfo(logAudio()) << " chunkSize: " << chunkSize;
qInfo(logAudio()) << " sampleRate: " << nativeSampleRate;
if (setup.isinput) {
err=Pa_OpenStream(&audio, &aParams, 0, this->nativeSampleRate, this->chunkSize, paNoFlag, &audioHandler::staticWrite, (void*)this);
}
else {
err=Pa_OpenStream(&audio, 0, &aParams, this->nativeSampleRate, this->chunkSize, paNoFlag, &audioHandler::staticRead, (void*)this);
}
if (err == paNoError) {
err = Pa_StartStream(audio);
}
if (err == paNoError) {
isInitialized = true;
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "device successfully opened";
}
else {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "failed to open device" << Pa_GetErrorText(err);
}
#else
format.setSampleSize(16);
format.setChannelCount(2);
format.setSampleRate(INTERNAL_SAMPLE_RATE);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::SignedInt);
if (setup.port.isNull())
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "No audio device was found. You probably need to install libqt5multimedia-plugins.";
return false;
}
else if (!setup.port.isFormatSupported(format))
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Format not supported, choosing nearest supported format - which may not work!";
format=setup.port.nearestFormat(format);
}
if (format.channelCount() > 2) {
format.setChannelCount(2);
}
else if (format.channelCount() < 1)
{
qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "No channels found, aborting setup.";
return false;
}
devChannels = format.channelCount();
nativeSampleRate = format.sampleRate();
// chunk size is always relative to Internal Sample Rate.
this->chunkSize = (nativeSampleRate / 50);
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Internal: sample rate" << format.sampleRate() << "channel count" << format.channelCount();
// We "hopefully" now have a valid format that is supported so try connecting
if (setup.isinput) {
audioInput = new QAudioInput(setup.port, format, this);
connect(audioInput, SIGNAL(notify()), SLOT(notified()));
connect(audioInput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
isInitialized = true;
}
else {
audioOutput = new QAudioOutput(setup.port, format, this);
#ifdef Q_OS_MAC
audioOutput->setBufferSize(chunkSize*4);
#endif
connect(audioOutput, SIGNAL(notify()), SLOT(notified()));
connect(audioOutput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
isInitialized = true;
}
#endif
// Setup resampler and opus if they are needed.
int resample_error = 0;
int opus_err = 0;
if (setup.isinput) {
resampler = wf_resampler_init(devChannels, nativeSampleRate, setup.samplerate, setup.resampleQuality, &resample_error);
if (setup.codec == 0x40 || setup.codec == 0x80) {
// Opus codec
encoder = opus_encoder_create(setup.samplerate, setup.radioChan, OPUS_APPLICATION_AUDIO, &opus_err);
opus_encoder_ctl(encoder, OPUS_SET_LSB_DEPTH(16));
opus_encoder_ctl(encoder, OPUS_SET_INBAND_FEC(1));
opus_encoder_ctl(encoder, OPUS_SET_DTX(1));
opus_encoder_ctl(encoder, OPUS_SET_PACKET_LOSS_PERC(5));
qInfo(logAudio()) << "Creating opus encoder: " << opus_strerror(opus_err);
}
}
else {
resampler = wf_resampler_init(devChannels, setup.samplerate, this->nativeSampleRate, setup.resampleQuality, &resample_error);
if (setup.codec == 0x40 || setup.codec == 0x80) {
// Opus codec
decoder = opus_decoder_create(setup.samplerate, setup.radioChan, &opus_err);
qInfo(logAudio()) << "Creating opus decoder: " << opus_strerror(opus_err);
}
}
unsigned int ratioNum;
unsigned int ratioDen;
wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
resampleRatio = static_cast<double>(ratioDen) / ratioNum;
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " resampleRatio: " << resampleRatio;
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "thread id" << QThread::currentThreadId();
#if !defined (RTAUDIO) && !defined(PORTAUDIO)
if (isInitialized) {
this->start();
}
#endif
return isInitialized;
}
#if !defined (RTAUDIO) && !defined(PORTAUDIO)
void audioHandler::start()
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "start() running";
if ((audioOutput == Q_NULLPTR || audioOutput->state() != QAudio::StoppedState) &&
(audioInput == Q_NULLPTR || audioInput->state() != QAudio::StoppedState)) {
return;
}
if (setup.isinput) {
#ifndef Q_OS_WIN
this->open(QIODevice::WriteOnly);
#else
this->open(QIODevice::WriteOnly | QIODevice::Unbuffered);
#endif
audioInput->start(this);
}
else {
#ifndef Q_OS_WIN
this->open(QIODevice::ReadOnly);
#else
this->open(QIODevice::ReadOnly | QIODevice::Unbuffered);
#endif
audioOutput->start(this);
}
}
#endif
void audioHandler::setVolume(unsigned char volume)
{
//this->volume = (qreal)volume/255.0;
this->volume = audiopot[volume];
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "setVolume: " << volume << "(" << this->volume << ")";
}
/// <summary>
/// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data
/// </summary>
/// <param name="data"></param>
/// <param name="maxlen"></param>
/// <returns></returns>
#if defined(RTAUDIO)
int audioHandler::readData(void* outputBuffer, void* inputBuffer,
unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
{
Q_UNUSED(inputBuffer);
Q_UNUSED(streamTime);
if (status == RTAUDIO_OUTPUT_UNDERFLOW)
qDebug(logAudio()) << "Underflow detected";
int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample and 2 channels
quint8* buffer = (quint8*)outputBuffer;
#elif defined(PORTAUDIO)
int audioHandler::readData(const void* inputBuffer, void* outputBuffer,
unsigned long nFrames, const PaStreamCallbackTimeInfo * streamTime, PaStreamCallbackFlags status)
{
Q_UNUSED(inputBuffer);
Q_UNUSED(streamTime);
Q_UNUSED(status);
int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample and 2 channels
quint8* buffer = (quint8*)outputBuffer;
#else
qint64 audioHandler::readData(char* buffer, qint64 nBytes)
{
#endif
// Calculate output length, always full samples
int sentlen = 0;
if (!isReady) {
isReady = true;
}
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(logAudio()) << "No more data available but buffer is not full! sentlen:" << sentlen << " nBytes:" << nBytes ;
break;
}
currentLatency = packet.time.msecsTo(QTime::currentTime());
// This shouldn't be required but if we did output a partial packet
// This will add the remaining packet data to the output buffer.
if (tempBuf.sent != tempBuf.data.length())
{
int send = qMin((int)nBytes - sentlen, tempBuf.data.length() - tempBuf.sent);
memcpy(buffer + sentlen, tempBuf.data.constData() + tempBuf.sent, send);
tempBuf.sent = tempBuf.sent + send;
sentlen = sentlen + send;
if (tempBuf.sent != tempBuf.data.length())
{
// We still don't have enough buffer space for this?
break;
}
//qDebug(logAudio()) << "Adding partial:" << send;
}
if (currentLatency > setup.latency) {
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Packet " << hex << packet.seq <<
" arrived too late (increase output latency!) " <<
dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
while (currentLatency > setup.latency/2) {
if (!ringBuf->try_read(packet)) {
break;
}
currentLatency = packet.time.msecsTo(QTime::currentTime());
}
}
int send = qMin((int)nBytes - sentlen, packet.data.length());
memcpy(buffer + sentlen, packet.data.constData(), send);
sentlen = sentlen + send;
if (send < 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;
}
/*
if (packet.seq <= lastSeq) {
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Duplicate/early audio packet: " << hex << lastSeq << " got " << hex << packet.seq;
}
else if (packet.seq != lastSeq + 1) {
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet.seq - 1;
}
*/
lastSeq = packet.seq;
}
}
//qDebug(logAudio()) << "looking for: " << nBytes << " got: " << sentlen;
// fill the rest of the buffer with silence
if (nBytes > sentlen) {
memset(buffer+sentlen,0,nBytes-sentlen);
}
#if defined(RTAUDIO)
return 0;
#elif defined(PORTAUDIO)
return 0;
#else
return nBytes;
#endif
}
#if defined(RTAUDIO)
int audioHandler::writeData(void* outputBuffer, void* inputBuffer,
unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
{
Q_UNUSED(outputBuffer);
Q_UNUSED(streamTime);
Q_UNUSED(status);
int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample and 2 channels
const char* data = (const char*)inputBuffer;
#elif defined(PORTAUDIO)
int audioHandler::writeData(const void* inputBuffer, void* outputBuffer,
unsigned long nFrames, const PaStreamCallbackTimeInfo * streamTime,
PaStreamCallbackFlags status)
{
Q_UNUSED(outputBuffer);
Q_UNUSED(streamTime);
Q_UNUSED(status);
int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample and 2 channels
const char* data = (const char*)inputBuffer;
#else
qint64 audioHandler::writeData(const char* data, qint64 nBytes)
{
#endif
if (!isReady) {
isReady = true;
}
int sentlen = 0;
//qDebug(logAudio()) << "nFrames" << nFrames << "nBytes" << nBytes;
int chunkBytes = chunkSize * devChannels * 2;
while (sentlen < nBytes) {
if (tempBuf.sent != chunkBytes)
{
int send = qMin((int)(nBytes - sentlen), chunkBytes - tempBuf.sent);
tempBuf.data.append(QByteArray::fromRawData(data + sentlen, send));
sentlen = sentlen + send;
tempBuf.seq = 0; // Not used in TX
tempBuf.time = QTime::currentTime();
tempBuf.sent = tempBuf.sent + send;
}
else {
//ringBuf->write(tempBuf);
if (!ringBuf->try_write(tempBuf))
{
qDebug(logAudio()) << "outgoing audio buffer full!";
break;
}
tempBuf.data.clear();
tempBuf.sent = 0;
}
}
//qDebug(logAudio()) << "sentlen" << sentlen;
#if defined(RTAUDIO)
return 0;
#elif defined(PORTAUDIO)
return 0;
#else
return nBytes;
#endif
}
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 (!isInitialized && !isReady)
{
qDebug(logAudio()) << "Packet received when stream was not ready";
return;
}
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);
if (nSamples != setup.samplerate / 50)
{
qInfo(logAudio()) << "Opus nSamples=" << nSamples << " expected:" << (setup.samplerate / 50);
return;
}
nSamples = opus_decode(decoder, in, inPacket.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();
return;
}
else {
if (int(nSamples * sizeof(qint16) * setup.radioChan) != outPacket.size())
{
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.
}
}
//qDebug(logAudio()) << "Got" << setup.bits << "bits, length" << inPacket.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);
qint16* out = (qint16*)outPacket.data();
for (int f = 0; f < inPacket.data.length(); f++)
{
int samp = (quint8)inPacket.data[f];
for (int g = setup.radioChan; g <= devChannels; g++)
{
if (setup.ulaw)
*out++ = ulaw_decode[samp] * this->volume;
else
*out++ = (qint16)((samp - 128) << 8) * this->volume;
}
}
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 (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();
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 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++)
{
*in = *in * this->volume;
in++;
}
}
}
/* 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 (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);
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;
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
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.
}
//qDebug(logAudio()) << "Adding packet to buffer:" << inPacket.seq << ": " << inPacket.data.length();
lastSentSeq = inPacket.seq;
if (!ringBuf->try_write(inPacket))
{
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size();
}
return;
}
void audioHandler::changeLatency(const quint16 newSize)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << setup.latency;
setup.latency = newSize;
delete ringBuf;
ringBuf = new wilt::Ring<audioPacket>(setup.latency / 8 + 1); // Should be customizable.
}
int audioHandler::getLatency()
{
return currentLatency;
}
void audioHandler::getNextAudioChunk(QByteArray& ret)
{
audioPacket packet;
packet.sent = 0;
if (isInitialized && ringBuf != Q_NULLPTR && ringBuf->try_read(packet))
{
currentLatency = packet.time.msecsTo(QTime::currentTime());
if (currentLatency > setup.latency) {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Packet " << hex << packet.seq <<
" arrived too late (increase output latency!) " <<
dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
// if (!ringBuf->try_read(packet))
// break;
// currentLatency = packet.time.msecsTo(QTime::currentTime());
}
//qDebug(logAudio) << "Chunksize" << this->chunkSize << "Packet size" << packet.data.length();
// Packet will arrive as stereo interleaved 16bit 48K
if (resampleRatio != 1.0)
{
quint32 outFrames = ((packet.data.length() / 2 / devChannels) * resampleRatio);
quint32 inFrames = (packet.data.length() / 2 / devChannels);
QByteArray outPacket((int)outFrames * 2 * devChannels, (char)0xff);
const qint16* in = (qint16*)packet.data.constData();
qint16* out = (qint16*)outPacket.data();
int err = 0;
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
if (err) {
qInfo(logAudio()) << (setup.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 = 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 (setup.radioChan == 1 && devChannels > 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();
if (setup.codec == 0x40 || setup.codec == 0x80)
{
//Are we using the opus codec?
qint16* in = (qint16*)packet.data.data();
/* Encode the frame. */
QByteArray outPacket(1275, (char)0xff); // Preset the output buffer size to MAXIMUM possible Opus frame size
unsigned char* out = (unsigned char*)outPacket.data();
int nbBytes = opus_encode(encoder, in, (setup.samplerate / 50), out, outPacket.length());
if (nbBytes < 0)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus encode failed:" << opus_strerror(nbBytes);
return;
}
else {
outPacket.resize(nbBytes);
packet.data.clear();
packet.data = outPacket; // Replace incoming data with converted.
}
}
else if (setup.bits == 8)
{
// Do we need to convert 16-bit to 8-bit?
QByteArray outPacket((int)packet.data.length() / 2, (char)0xff);
qint16* in = (qint16*)packet.data.data();
for (int f = 0; f < outPacket.length(); f++)
{
qint16 sample = *in++;
if (setup.ulaw) {
int sign = (sample >> 8) & 0x80;
if (sign)
sample = (short)-sample;
if (sample > cClip)
sample = cClip;
sample = (short)(sample + cBias);
int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
int mantissa = (sample >> (exponent + 3)) & 0x0F;
int compressedByte = ~(sign | (exponent << 4) | mantissa);
outPacket[f] = (quint8)compressedByte;
}
else {
int compressedByte = (((sample + 32768) >> 8) & 0xff);
outPacket[f] = (quint8)compressedByte;
}
}
packet.data.clear();
packet.data = outPacket; // Copy output packet back to input buffer.
}
ret = packet.data;
//qDebug(logAudio()) << "Now radio format, length" << packet.data.length();
}
return;
}
#if !defined (RTAUDIO) && !defined(PORTAUDIO)
qint64 audioHandler::bytesAvailable() const
{
return 0;
}
bool audioHandler::isSequential() const
{
return true;
}
void audioHandler::notified()
{
}
void audioHandler::stateChanged(QAudio::State state)
{
// Process the state
switch (state)
{
case QAudio::IdleState:
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio now in idle state: " << audioBuffer.size() << " packets in buffer";
if (audioOutput != Q_NULLPTR && audioOutput->error() == QAudio::UnderrunError)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "buffer underrun";
//audioOutput->suspend();
}
break;
}
case QAudio::ActiveState:
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio now in active state: " << audioBuffer.size() << " packets in buffer";
break;
}
case QAudio::SuspendedState:
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio now in suspended state: " << audioBuffer.size() << " packets in buffer";
break;
}
case QAudio::StoppedState:
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio now in stopped state: " << audioBuffer.size() << " packets in buffer";
break;
}
default: {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unhandled audio state: " << audioBuffer.size() << " packets in buffer";
}
}
}
void audioHandler::stop()
{
if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) {
// Stop audio output
audioOutput->stop();
this->stop();
this->close();
delete audioOutput;
audioOutput = Q_NULLPTR;
}
if (audioInput != Q_NULLPTR && audioInput->state() != QAudio::StoppedState) {
// Stop audio output
audioInput->stop();
this->stop();
this->close();
delete audioInput;
audioInput = Q_NULLPTR;
}
isInitialized = false;
}
#endif