/*
	This class handles both RX and TX audio, each is created as a separate 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"



audioHandler::audioHandler(QObject* parent) 
{
	Q_UNUSED(parent)
}

audioHandler::~audioHandler()
{

	if (isInitialized) {
		stop();
	}

	if (ringBuf != Q_NULLPTR) {
		delete ringBuf;
	}

	if (audioInput != Q_NULLPTR) {
		audioInput = Q_NULLPTR;
		delete audioInput;
	}
	if (audioOutput != Q_NULLPTR) {
		delete audioOutput;
		audioOutput = Q_NULLPTR;
	}


	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.format.setChannelCount(1);
	setup.format.setSampleSize(8);
	setup.format.setSampleType(QAudioFormat::UnSignedInt);
	qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "audio handler starting:" << setup.name;

	if (setup.codec == 0x01 || setup.codec == 0x20) {
		setup.ulaw = true;
	}
	if (setup.codec == 0x08 || setup.codec == 0x10 || setup.codec == 0x20 || setup.codec == 0x80) {
		setup.format.setChannelCount(2);
	}
	if (setup.codec == 0x04 || setup.codec == 0x10 || setup.codec == 0x40 || setup.codec == 0x80) {
		setup.format.setSampleSize(16);
		setup.format.setSampleType(QAudioFormat::SignedInt);
	}

	qDebug(logAudio()) << "Creating" << (setup.isinput ? "Input" : "Output") << "audio device:" << setup.name <<
		", bits" << setup.format.sampleSize() <<
		", codec" << setup.codec <<
		", latency" << setup.latency <<
		", localAFGain" << setup.localAFgain <<
		", radioChan" << setup.format.channelCount() <<
		", resampleQuality" << setup.resampleQuality <<
		", samplerate" << setup.format.sampleRate() <<
		", uLaw" << setup.ulaw;


	tempBuf.sent = 0;

    if(!setup.isinput)
    {
        this->setVolume(setup.localAFgain);
    }

	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();

	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()));
		isInitialized = true;
	}
	else {
		audioOutput = new QAudioOutput(setup.port, format, this);

		audioOutput->setBufferSize(getAudioSize(setup.latency, format));
		isInitialized = true;
	}

	// 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.format.sampleRate(), setup.resampleQuality, &resample_error);
		if (setup.codec == 0x40 || setup.codec == 0x80) {
			// Opus codec
			encoder = opus_encoder_create(setup.format.sampleRate(), setup.format.channelCount(), 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 {

		//resampBufs = new r8b::CFixedBuffer<double>[format.channelCount()];
		//resamps = new r8b::CPtrKeeper<r8b::CDSPResampler24*>[format.channelCount()];
		resampler = wf_resampler_init(devChannels, setup.format.sampleRate(), this->nativeSampleRate, setup.resampleQuality, &resample_error);
		if (setup.codec == 0x40 || setup.codec == 0x80) {
			// Opus codec
			decoder = opus_decoder_create(setup.format.sampleRate(), setup.format.sampleRate(), &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();

	this->start();
	return isInitialized;
}

void audioHandler::start()
{
	qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "start() running";

	if (setup.isinput) {
		audioDevice = audioInput->start();
		connect(audioInput, &QAudioOutput::destroyed, audioDevice, &QIODevice::deleteLater, Qt::UniqueConnection);
		connect(audioDevice, &QIODevice::destroyed, this, &QAudioInput::deleteLater, Qt::UniqueConnection);
	}
	else {
		audioDevice = audioOutput->start();
		connect(audioOutput, &QAudioOutput::destroyed, audioDevice, &QIODevice::deleteLater, Qt::UniqueConnection);
		connect(audioDevice, &QIODevice::destroyed, this, &QAudioOutput::deleteLater, Qt::UniqueConnection);
	}
	if (!audioDevice) {
		qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio device failed to start()";
		return;
	}
}

void audioHandler::setVolume(unsigned char volume)
{
    this->volume = audiopot[volume];
	qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "setVolume: " << volume << "(" << this->volume << ")";
}


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)

	audioPacket livePacket = inPacket;

	if (setup.codec == 0x40 || setup.codec == 0x80) {
		/* Opus data */
		unsigned char* in = (unsigned char*)inPacket.data.data();

		/* Decode the frame. */
		QByteArray outPacket((setup.format.sampleRate() / 50) *  sizeof(qint16) * setup.format.channelCount(), (char)0xff); // Preset the output buffer size.
		qint16* out = (qint16*)outPacket.data();
		int nSamples = opus_packet_get_nb_samples(in, livePacket.data.size(),setup.format.sampleRate());
		if (nSamples == -1) {
			// No opus data yet?
			return;
		} 
		else if (nSamples != setup.format.sampleRate() / 50)
		{
			qDebug(logAudio()) << "Opus nSamples=" << nSamples << " expected:" << (setup.format.sampleRate() / 50);
			return;
		}
		if (livePacket.seq > lastSentSeq + 1) {
			nSamples = opus_decode(decoder, in, livePacket.data.size(), out, (setup.format.sampleRate() / 50), 1);
		}
		else {
			nSamples = opus_decode(decoder, in, livePacket.data.size(), out, (setup.format.sampleRate() / 50), 0);
		}
		if (nSamples < 0)
		{
			qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decode failed:" << opus_strerror(nSamples) << "packet size" << livePacket.data.length();
			return;
		}
		else {
			if (int(nSamples * sizeof(qint16) * setup.format.channelCount()) != outPacket.size())
			{
				qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decoder mismatch: nBytes:" << nSamples * sizeof(qint16) * setup.format.channelCount() << "outPacket:" << outPacket.size();
				outPacket.resize(nSamples * sizeof(qint16) * setup.format.channelCount());
			}
			//qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decoded" << livePacket.data.size() << "bytes, into" << outPacket.length() << "bytes";
			livePacket.data.clear();
			livePacket.data = outPacket; // Replace incoming data with converted.
		}
	}

	// Process uLaw.
	if (setup.ulaw)
	{
		// Current packet is 8bit so need to create a new buffer that is 16bit 
		QByteArray outPacket((int)livePacket.data.length() * 2, (char)0xff);
		qint16* out = (qint16*)outPacket.data();
		for (int f = 0; f < livePacket.data.length(); f++)
		{
			*out++ = ulaw_decode[(quint8)livePacket.data[f]];
		}
		livePacket.data.clear();
		livePacket.data = outPacket; // Replace incoming data with converted.
		setup.format.setSampleSize(16);
		setup.format.setSampleType(QAudioFormat::SignedInt);
		// Buffer now contains 16bit signed samples.
	}



	if (!livePacket.data.isEmpty()) {

		Eigen::VectorXf samplesF;
		if (setup.format.sampleSize() == 16)
		{
			VectorXint16 samplesI = Eigen::Map<VectorXint16>(reinterpret_cast<qint16*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint16)));
			samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint16>::max());
		}
		else
		{
			VectorXuint8 samplesI = Eigen::Map<VectorXuint8>(reinterpret_cast<quint8*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(quint8)));
			samplesF = samplesI.cast<float>() / float(std::numeric_limits<quint8>::max());;
		}

		// Set the max amplitude found in the vector
		amplitude = samplesF.array().abs().maxCoeff();
		// Set the volume
		samplesF *= volume;

		// Convert mono to stereo
		if (setup.format.channelCount() == 1) {
			Eigen::VectorXf samplesTemp(samplesF.size() * 2);
			Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data(), samplesF.size()) = samplesF;
			Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data() + 1, samplesF.size()) = samplesF;
			samplesF = samplesTemp;
		}


		if (format.sampleType() == QAudioFormat::SignedInt)
		{
			Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
			VectorXint16 samplesI = samplesITemp.cast<qint16>();
			livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
		}
		else
		{
			livePacket.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
		}


		if (resampleRatio != 1.0) {

			// We need to resample
			// We have a stereo 16bit stream.
			quint32 outFrames = ((livePacket.data.length() / 2 / devChannels) * resampleRatio);
			quint32 inFrames = (livePacket.data.length() / 2 / devChannels);
			QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.

			const qint16* in = (qint16*)livePacket.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;
			}
			livePacket.data.clear();
			livePacket.data = outPacket; // Replace incoming data with converted.
		}


		//qDebug(logAudio()) << "Adding packet to buffer:" << livePacket.seq << ": " << livePacket.data.length();
		
		currentLatency = livePacket.time.msecsTo(QTime::currentTime()) + getAudioDuration(audioOutput->bufferSize()-audioOutput->bytesFree(),format);

		if (audioDevice != Q_NULLPTR) {
			audioDevice->write(livePacket.data);
		}
		if ((inPacket.seq > lastSentSeq + 1) && (setup.codec == 0x40 || setup.codec == 0x80)) {
			qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Attempting FEC on packet" << inPacket.seq << "as last is" << lastSentSeq;
			lastSentSeq = inPacket.seq;
			incomingAudio(inPacket); // Call myself again to run the packet a second time (FEC)
		}

		lastSentSeq = inPacket.seq;
	}

	return;
}

void audioHandler::changeLatency(const quint16 newSize)
{

	qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << setup.latency;
	setup.latency = newSize;

	if (!setup.isinput) {
		stop();
		audioOutput->setBufferSize(getAudioSize(setup.latency, format));
		start();
	}

}

int audioHandler::getLatency()
{
	return currentLatency;
}



void audioHandler::getNextAudioChunk(QByteArray& ret)
{
	audioPacket packet;
	packet.sent = 0;
	if (audioDevice != Q_NULLPTR) {
		packet.data = audioDevice->readAll();
	}
	if (packet.data.length() > 0)
	{		
		// 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;
			}
			packet.data.clear();
			packet.data = outPacket; // Copy output packet back to input buffer.
		}

		//qDebug(logAudio()) << "Now resampled, length" << packet.data.length();

		int tempAmplitude = 0;
		// Do we need to convert mono to stereo?
		if (setup.format.channelCount() == 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++)
			{
				tempAmplitude = qMax(tempAmplitude, (int)(abs(*in) / 256));
				*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.format.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.format.sampleSize() == 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;
				}
				tempAmplitude = qMax(tempAmplitude, abs(outPacket[f]));
			}
			packet.data.clear();
			packet.data = outPacket; // Copy output packet back to input buffer.
		} 
		amplitude = tempAmplitude;

		ret = packet.data;

	}

	return;

}


void audioHandler::stop()
{
	qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "stop() running";

	if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) {
		// Stop audio output
		audioOutput->stop();
	}

	if (audioInput != Q_NULLPTR && audioInput->state() != QAudio::StoppedState) {
		// Stop audio output
		audioInput->stop();
	}
	audioDevice = Q_NULLPTR;
}


quint16 audioHandler::getAmplitude()
{
	return static_cast<quint16>(amplitude * 255.0);
}