kopia lustrzana https://gitlab.com/eliggett/wfview
473 wiersze
15 KiB
C++
473 wiersze
15 KiB
C++
/*
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This class handles both RX and TX audio, each is created as a seperate instance of the class
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but as the setup/handling if output (RX) and input (TX) devices is so similar I have combined them.
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*/
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#include "audiohandler.h"
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#include "logcategories.h"
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#include "ulaw.h"
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audioHandler::audioHandler(QObject* parent) :
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isInitialized(false),
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isUlaw(false),
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audioLatency(0),
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isInput(0),
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chunkAvailable(false)
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{
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Q_UNUSED(parent)
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}
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audioHandler::~audioHandler()
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{
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//stop();
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if (resampler != Q_NULLPTR) {
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speex_resampler_destroy(resampler);
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}
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if (audio.isStreamRunning())
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{
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audio.stopStream();
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audio.closeStream();
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}
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if (ringBuf != Q_NULLPTR)
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delete ringBuf;
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}
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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)
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{
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if (isInitialized) {
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return false;
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}
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this->audioLatency = latency;
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this->isUlaw = ulaw;
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this->isInput = isinput;
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this->radioSampleBits = bits;
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this->radioSampleRate = samplerate;
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this->radioChannels = radioChan;
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// chunk size is always relative to Internal Sample Rate.
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ringBuf = new wilt::Ring<audioPacket>(100); // Should be customizable.
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tempBuf.sent = 0;
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if (port > 0) {
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aParams.deviceId = port;
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}
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else if (isInput) {
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aParams.deviceId = audio.getDefaultInputDevice();
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}
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else {
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aParams.deviceId = audio.getDefaultOutputDevice();
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}
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aParams.firstChannel = 0;
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try {
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info = audio.getDeviceInfo(aParams.deviceId);
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}
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catch (RtAudioError& e) {
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qInfo(logAudio()) << "Device error:" << aParams.deviceId << ":" << QString::fromStdString(e.getMessage());
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return false;
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}
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if (info.probed)
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{
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// Per channel chunk size.
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aParams.nChannels = 2; // Internally this is always 2 channels for TX and RX.
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this->chunkSize = (info.preferredSampleRate / 50);
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed";
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if (info.nativeFormats == 0)
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{
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qInfo(logAudio()) << " No natively supported data formats!";
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return false;
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}
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else {
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qDebug(logAudio()) << " Supported formats:" <<
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(info.nativeFormats & RTAUDIO_SINT8 ? "8-bit int," : "") <<
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(info.nativeFormats & RTAUDIO_SINT16 ? "16-bit int," : "") <<
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(info.nativeFormats & RTAUDIO_SINT24 ? "24-bit int," : "") <<
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(info.nativeFormats & RTAUDIO_SINT32 ? "32-bit int," : "") <<
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(info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") <<
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(info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : "");
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qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate;
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if (isInput) {
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devChannels = info.inputChannels;
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}
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else {
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devChannels = info.outputChannels;
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}
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qInfo(logAudio()) << " Channels:" << devChannels;
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if (devChannels > 2) {
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devChannels = 2;
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}
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aParams.nChannels = devChannels;
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}
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qInfo(logAudio()) << " chunkSize: " << chunkSize;
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}
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else
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{
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qCritical(logAudio()) << (isInput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") could not be probed, check audio configuration!";
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return false;
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}
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int resample_error = 0;
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if (isInput) {
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resampler = wf_resampler_init(devChannels, info.preferredSampleRate, samplerate, resampleQuality, &resample_error);
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try {
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audio.openStream(NULL, &aParams, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticWrite, this);
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audio.startStream();
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}
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catch (RtAudioError& e) {
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qInfo(logAudio()) << "Error opening:" << QString::fromStdString(e.getMessage());
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return false;
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}
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}
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else
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{
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resampler = wf_resampler_init(devChannels, samplerate, info.preferredSampleRate, resampleQuality, &resample_error);
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try {
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audio.openStream(&aParams, NULL, RTAUDIO_SINT16, info.preferredSampleRate, &this->chunkSize, &staticRead, this);
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audio.startStream();
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}
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catch (RtAudioError& e) {
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qInfo(logAudio()) << "Error opening:" << QString::fromStdString(e.getMessage());
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return false;
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}
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}
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "device successfully opened";
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "detected latency:" <<audio.getStreamLatency();
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wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen;
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "thread id" << QThread::currentThreadId();
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return isInitialized;
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}
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void audioHandler::setVolume(unsigned char volume)
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{
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this->volume = (qreal)volume/255.0;
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << this->volume << ")";
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}
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/// <summary>
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/// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data
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/// </summary>
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/// <param name="data"></param>
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/// <param name="maxlen"></param>
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/// <returns></returns>
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int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
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{
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Q_UNUSED(inputBuffer);
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Q_UNUSED(streamTime);
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// Calculate output length, always full samples
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int sentlen = 0;
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quint8* buffer = (quint8*)outputBuffer;
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if (status == RTAUDIO_OUTPUT_UNDERFLOW)
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qDebug(logAudio()) << "Underflow detected";
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int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample and 2 channels
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if (ringBuf->size()>0)
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{
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// Output buffer is ALWAYS 16 bit.
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//qDebug(logAudio()) << "Read: nFrames" << nFrames << "nBytes" << nBytes;
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while (sentlen < nBytes)
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{
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audioPacket packet;
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if (!ringBuf->try_read(packet))
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{
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qDebug() << "No more data available but buffer is not full! sentlen:" << sentlen << " nBytes:" << nBytes ;
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break;
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}
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currentLatency = packet.time.msecsTo(QTime::currentTime());
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// This shouldn't be required but if we did output a partial packet
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// This will add the remaining packet data to the output buffer.
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if (tempBuf.sent != tempBuf.data.length())
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{
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int send = qMin((int)nBytes - sentlen, tempBuf.data.length() - tempBuf.sent);
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memcpy(buffer + sentlen, tempBuf.data.constData() + tempBuf.sent, send);
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tempBuf.sent = tempBuf.sent + send;
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sentlen = sentlen + send;
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if (tempBuf.sent != tempBuf.data.length())
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{
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// We still don't have enough buffer space for this?
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break;
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}
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//qDebug(logAudio()) << "Adding partial:" << send;
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}
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if (currentLatency > (int)audioLatency) {
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Packet " << hex << packet.seq <<
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" arrived too late (increase output latency!) " <<
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dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
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lastSeq = packet.seq;
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if (!ringBuf->try_read(packet))
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break;
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currentLatency = packet.time.msecsTo(QTime::currentTime());
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}
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int send = qMin((int)nBytes - sentlen, packet.data.length());
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memcpy(buffer + sentlen, packet.data.constData(), send);
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sentlen = sentlen + send;
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if (send < packet.data.length())
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{
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//qDebug(logAudio()) << "Asking for partial, sent:" << send << "packet length" << packet.data.length();
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tempBuf = packet;
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tempBuf.sent = tempBuf.sent + send;
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lastSeq = packet.seq;
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break;
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}
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if (packet.seq <= lastSeq) {
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qDebug(logAudio()) << (isInput ? "Input" : "Output") << "Duplicate/early audio packet: " << hex << lastSeq << " got " << hex << packet.seq;
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}
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else if (packet.seq != lastSeq + 1) {
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qDebug(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet.seq - 1;
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}
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lastSeq = packet.seq;
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}
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}
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//qDebug(logAudio()) << "looking for: " << nBytes << " got: " << sentlen;
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// fill the rest of the buffer with silence
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if (nBytes > sentlen) {
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memset(buffer+sentlen,0,nBytes-sentlen);
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}
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return 0;
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}
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int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
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{
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Q_UNUSED(outputBuffer);
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Q_UNUSED(streamTime);
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Q_UNUSED(status);
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int sentlen = 0;
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int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample
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const char* data = (const char*)inputBuffer;
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//qDebug(logAudio()) << "nFrames" << nFrames << "nBytes" << nBytes;
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while (sentlen < nBytes) {
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if (tempBuf.sent != nBytes)
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{
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int send = qMin((int)(nBytes - sentlen), (int)nBytes - tempBuf.sent);
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tempBuf.data.append(QByteArray::fromRawData(data + sentlen, send));
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sentlen = sentlen + send;
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tempBuf.seq = 0; // Not used in TX
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tempBuf.time = QTime::currentTime();
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tempBuf.sent = tempBuf.sent + send;
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}
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else {
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if (!ringBuf->try_write(tempBuf))
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{
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qDebug(logAudio()) << "outgoing audio buffer full!";
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break;
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}
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tempBuf.data.clear();
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tempBuf.sent = 0;
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}
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}
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//qDebug(logAudio()) << "sentlen" << sentlen;
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return 0;
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}
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void audioHandler::incomingAudio(audioPacket inPacket)
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{
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// No point buffering audio until stream is actually running.
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// Regardless of the radio stream format, the buffered audio will ALWAYS be
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// 16bit sample interleaved stereo 48K (or whatever the native sample rate is)
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if (!audio.isStreamRunning())
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{
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qDebug(logAudio()) << "Packet received before stream was started";
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return;
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}
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//qDebug(logAudio()) << "Got" << radioSampleBits << "bits, length" << inPacket.data.length();
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// Incoming data is 8bits?
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if (radioSampleBits == 8)
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{
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// Current packet is 8bit so need to create a new buffer that is 16bit
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QByteArray outPacket((int)inPacket.data.length() * 2 *(devChannels/radioChannels), (char)0xff);
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qint16* out = (qint16*)outPacket.data();
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for (int f = 0; f < inPacket.data.length(); f++)
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{
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for (int g = radioChannels; g <= devChannels; g++)
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{
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if (isUlaw)
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*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
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else
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*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640 * this->volume);
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}
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}
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inPacket.data.clear();
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inPacket.data = outPacket; // Replace incoming data with converted.
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}
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else
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{
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// This is already a 16bit stream, do we need to convert to stereo?
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if (radioChannels == 1 && devChannels > 1) {
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// Yes
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QByteArray outPacket(inPacket.data.length() * 2, (char)0xff); // Preset the output buffer size.
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qint16* in = (qint16*)inPacket.data.data();
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qint16* out = (qint16*)outPacket.data();
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for (int f = 0; f < inPacket.data.length() / 2; f++)
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{
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*out++ = (qint16)*in * this->volume;
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*out++ = (qint16)*in++ * this->volume;
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}
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inPacket.data.clear();
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inPacket.data = outPacket; // Replace incoming data with converted.
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}
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else
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{
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// We already have the same number of channels so just update volume.
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qint16* in = (qint16*)inPacket.data.data();
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for (int f = 0; f < inPacket.data.length() / 2; f++)
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{
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*in = *in * this->volume;
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in++;
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}
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}
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}
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/* We now have an array of 16bit samples in the NATIVE samplerate of the radio
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If the radio sample rate is below 48000, we need to resample.
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*/
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//qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
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if (ratioDen != 1) {
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// We need to resample
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// We have a stereo 16bit stream.
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quint32 outFrames = ((inPacket.data.length() / 2 / devChannels) * ratioDen);
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quint32 inFrames = (inPacket.data.length() / 2 / devChannels);
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QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.
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const qint16* in = (qint16*)inPacket.data.constData();
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qint16* out = (qint16*)outPacket.data();
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int err = 0;
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err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
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if (err) {
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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inPacket.data.clear();
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inPacket.data = outPacket; // Replace incoming data with converted.
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}
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//qDebug(logAudio()) << "Adding packet to buffer:" << inPacket.seq << ": " << inPacket.data.length();
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if (!ringBuf->try_write(inPacket))
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{
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qDebug(logAudio()) << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size();
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}
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return;
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}
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void audioHandler::changeLatency(const quint16 newSize)
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{
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << audioLatency;
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audioLatency = newSize;
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}
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int audioHandler::getLatency()
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{
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return currentLatency;
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}
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void audioHandler::getNextAudioChunk(QByteArray& ret)
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{
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audioPacket packet;
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packet.sent = 0;
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if (ringBuf != Q_NULLPTR && ringBuf->try_read(packet))
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{
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//qDebug(logAudio) << "Chunksize" << this->chunkSize << "Packet size" << packet.data.length();
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// Packet will arrive as stereo interleaved 16bit 48K
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if (ratioNum != 1)
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{
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quint32 outFrames = ((packet.data.length() / 2 / devChannels) / ratioNum);
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quint32 inFrames = (packet.data.length() / 2 / devChannels);
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QByteArray outPacket((int)outFrames * 2 * devChannels, (char)0xff);
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const qint16* in = (qint16*)packet.data.constData();
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qint16* out = (qint16*)outPacket.data();
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int err = 0;
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err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
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if (err) {
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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//qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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//qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length();
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packet.data.clear();
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packet.data = outPacket; // Copy output packet back to input buffer.
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}
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//qDebug(logAudio()) << "Now resampled, length" << packet.data.length();
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// Do we need to convert mono to stereo?
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if (radioChannels == 1 && devChannels > 1)
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{
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// Strip out right channel?
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QByteArray outPacket(packet.data.length()/2, (char)0xff);
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const qint16* in = (qint16*)packet.data.constData();
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qint16* out = (qint16*)outPacket.data();
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for (int f = 0; f < outPacket.length()/2; f++)
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{
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*out++ = *in++;
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in++; // Skip each even channel.
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}
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packet.data.clear();
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packet.data = outPacket; // Copy output packet back to input buffer.
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}
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//qDebug(logAudio()) << "Now mono, length" << packet.data.length();
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// Do we need to convert 16-bit to 8-bit?
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if (radioSampleBits == 8) {
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QByteArray outPacket((int)packet.data.length() / 2, (char)0xff);
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qint16* in = (qint16*)packet.data.data();
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for (int f = 0; f < outPacket.length(); f++)
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{
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quint8 outdata = 0;
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if (isUlaw) {
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qint16 enc = qFromLittleEndian<quint16>(in + f);
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if (enc >= 0)
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outdata = ulaw_encode[enc];
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else
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outdata = 0x7f & ulaw_encode[-enc];
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}
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else {
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outdata = (quint8)(((qFromLittleEndian<qint16>(in + f) >> 8) ^ 0x80) & 0xff);
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}
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outPacket[f] = (char)outdata;
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}
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packet.data.clear();
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packet.data = outPacket; // Copy output packet back to input buffer.
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}
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ret = packet.data;
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//qDebug(logAudio()) << "Now radio format, length" << packet.data.length();
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}
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return;
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}
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