kopia lustrzana https://gitlab.com/eliggett/wfview
422 wiersze
12 KiB
C++
422 wiersze
12 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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audioHandler::audioHandler(QObject* parent) :
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QIODevice(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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}
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audioHandler::~audioHandler()
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{
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//stop();
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if (resampler) {
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speex_resampler_destroy(resampler);
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}
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}
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bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16 samplerate, const quint16 latency, const bool ulaw, const bool isinput, QString port, int device, 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->audioDevice = device;
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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 = channels;
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// chunk size is always relative to Internal Sample Rate.
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this->chunkSize = (INTERNAL_SAMPLE_RATE / 25) * radioChannels;
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if (device != 0) {
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aParams.deviceId = device;
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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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info = audio.getDeviceInfo(aParams.deviceId);
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if (info.probed)
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{
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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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qInfo(logAudio()) << " No natively supported data formats!";
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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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}
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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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}
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int resample_error = 0;
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if (isInput) {
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resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error);
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//audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &output, (void*)&data);
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}
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else
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{
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resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error);
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//audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &output, (void*)&data);
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}
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wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
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qInfo(logAudio()) << " wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen;
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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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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")";
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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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qint64 audioHandler::readData(char* data, qint64 maxlen)
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{
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// Calculate output length, always full samples
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int sentlen = 0;
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//qInfo(logAudio()) << "Looking for: " << maxlen << " bytes";
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// We must lock the mutex for the entire time that the buffer may be modified.
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// Get next packet from buffer.
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if (!audioBuffer.isEmpty())
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{
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// Output buffer is ALWAYS 16 bit.
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QMutexLocker locker(&mutex);
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auto packet = audioBuffer.begin();
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while (packet != audioBuffer.end() && sentlen < maxlen)
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{
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int timediff = packet->time.msecsTo(QTime::currentTime());
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if (timediff > (int)audioLatency * 2) {
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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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while (packet != audioBuffer.end() && timediff > (int)audioLatency) {
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timediff = packet->time.msecsTo(QTime::currentTime());
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lastSeq = packet->seq;
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packet = audioBuffer.erase(packet); // returns next packet
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}
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if (packet == audioBuffer.end()) {
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break;
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}
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}
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// If we got here then packet time must be within latency threshold
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if (packet->seq == lastSeq + 1 || packet->seq <= lastSeq)
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{
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int send = qMin((int)maxlen - sentlen, packet->dataout.length() - packet->sent);
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lastSeq = packet->seq;
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//qInfo(logAudio()) << "Packet " << hex << packet->seq << " arrived on time " << Qt::dec << packet->time.msecsTo(QTime::currentTime()) << "ms";
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memcpy(data + sentlen, packet->dataout.constData() + packet->sent, send);
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sentlen = sentlen + send;
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if (send == packet->dataout.length() - packet->sent)
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{
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//qInfo(logAudio()) << "Get next packet";
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packet = audioBuffer.erase(packet); // returns next packet
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}
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else
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{
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// Store sent amount (could be zero if audioOutput buffer full) then break.
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packet->sent = send;
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break;
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}
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}
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else {
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet->seq - 1;
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lastSeq = packet->seq;
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}
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}
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}
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else {
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// Fool audio system into thinking it has valid data, this seems to be required
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// for MacOS Built in audio but shouldn't cause any issues with other platforms.
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memset(data, 0x0, maxlen);
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return maxlen;
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}
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return sentlen;
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}
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qint64 audioHandler::writeData(const char* data, qint64 len)
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{
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qint64 sentlen = 0;
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QMutexLocker locker(&mutex);
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audioPacket* current;
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while (sentlen < len) {
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if (!audioBuffer.isEmpty())
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{
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if (audioBuffer.last().sent == chunkSize)
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{
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audioBuffer.append(audioPacket());
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audioBuffer.last().sent = 0;
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}
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}
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else
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{
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audioBuffer.append(audioPacket());
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audioBuffer.last().sent = 0;
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}
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current = &audioBuffer.last();
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int send = qMin((int)(len - sentlen), (int)chunkSize - current->sent);
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current->datain.append(QByteArray::fromRawData(data + sentlen, send));
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sentlen = sentlen + send;
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current->seq = 0; // Not used in TX
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current->time = QTime::currentTime();
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current->sent = current->datain.length();
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if (current->sent == chunkSize)
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{
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chunkAvailable = true;
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}
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else if (audioBuffer.length() <= 1 && current->sent != chunkSize) {
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chunkAvailable = false;
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}
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}
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return (sentlen); // Always return the same number as we received
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}
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qint64 audioHandler::bytesAvailable() const
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{
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return 0;
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}
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bool audioHandler::isSequential() const
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{
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return true;
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}
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void audioHandler::notified()
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{
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}
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void audioHandler::stateChanged(QAudio::State state)
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{
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}
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void audioHandler::incomingAudio(audioPacket data)
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{
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QMutexLocker locker(&mutex);
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// Incoming data is 8bits?
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if (radioSampleBits == 8)
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{
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QByteArray inPacket((int)data.datain.length() * 2, (char)0xff);
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qint16* in = (qint16*)inPacket.data();
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for (int f = 0; f < data.datain.length(); f++)
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{
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if (isUlaw)
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{
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in[f] = ulaw_decode[(quint8)data.datain[f]];
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}
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else
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{
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// Convert 8-bit sample to 16-bit
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in[f] = (qint16)(((quint8)data.datain[f] << 8) - 32640);
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}
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}
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data.datain = inPacket; // Replace incoming data with converted.
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}
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//qInfo(logAudio()) << "Adding packet to buffer:" << data.seq << ": " << data.datain.length();
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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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if (ratioDen != 1) {
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// We need to resample
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quint32 outFrames = ((data.datain.length() / 2) * ratioDen) / radioChannels;
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quint32 inFrames = (data.datain.length() / 2) / radioChannels;
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data.dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size.
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int err = 0;
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if (this->radioChannels == 1) {
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err = wf_resampler_process_int(resampler, 0, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames);
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}
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else {
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err = wf_resampler_process_interleaved_int(resampler, (const qint16*)data.datain.constData(), &inFrames, (qint16*)data.dataout.data(), &outFrames);
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}
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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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}
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else {
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data.dataout = data.datain;
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}
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audioBuffer.push_back(data);
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// Sort the buffer by seq number. This is important and audio packets may have arrived out-of-order
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std::sort(audioBuffer.begin(), audioBuffer.end(),
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[](const audioPacket& a, const audioPacket& b) -> bool
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{
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return a.seq < b.seq;
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});
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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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void audioHandler::getLatency()
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{
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emit sendLatency(audioLatency);
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}
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bool audioHandler::isChunkAvailable()
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{
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return (chunkAvailable);
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}
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void audioHandler::getNextAudioChunk(QByteArray& ret)
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{
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if (!audioBuffer.isEmpty() && chunkAvailable)
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{
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QMutexLocker locker(&mutex);
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// Skip through audio buffer deleting any old entry.
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auto packet = audioBuffer.begin();
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while (packet != audioBuffer.end())
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{
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if (packet->time.msecsTo(QTime::currentTime()) > 100) {
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//qInfo(logAudio()) << "TX Packet too old " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms";
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packet = audioBuffer.erase(packet); // returns next packet
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}
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else {
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if (packet->datain.length() == chunkSize && ret.length() == 0)
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{
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/* We now have an array of samples in the computer native format (48000)
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If the radio sample rate is below 48000, we need to resample.
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*/
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if (ratioNum != 1)
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{
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// We need to resample (we are STILL 16 bit!)
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quint32 outFrames = ((packet->datain.length() / 2) / ratioNum) / radioChannels;
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quint32 inFrames = (packet->datain.length() / 2) / radioChannels;
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packet->dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size.
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const qint16* in = (qint16*)packet->datain.constData();
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qint16* out = (qint16*)packet->dataout.data();
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int err = 0;
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if (this->radioChannels == 1) {
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err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames);
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}
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else {
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err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
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}
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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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if (radioSampleBits == 8)
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{
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packet->datain = packet->dataout; // Copy output packet back to input buffer.
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packet->dataout.clear(); // Buffer MUST be cleared ready to be re-filled by the upsampling below.
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}
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}
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else if (radioSampleBits == 16) {
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// Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise.
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packet->dataout = packet->datain;
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}
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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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packet->dataout.resize(packet->datain.length() / 2);
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qint16* in = (qint16*)packet->datain.data();
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for (int f = 0; f < packet->dataout.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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packet->dataout[f] = (char)outdata;
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}
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}
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ret = packet->dataout;
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packet = audioBuffer.erase(packet); // returns next packet
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}
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else {
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packet++;
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}
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}
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}
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}
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return;
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}
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