kopia lustrzana https://gitlab.com/eliggett/wfview
314 wiersze
14 KiB
C++
314 wiersze
14 KiB
C++
#include "audioconverter.h"
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#include "logcategories.h"
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#include "ulaw.h"
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audioConverter::audioConverter(QObject* parent) : QObject(parent)
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{
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}
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bool audioConverter::init(QAudioFormat inFormat, QAudioFormat outFormat, quint8 opusComplexity, quint8 resampleQuality)
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{
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this->inFormat = inFormat;
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this->outFormat = outFormat;
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this->opusComplexity = opusComplexity;
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this->resampleQuality = resampleQuality;
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qInfo(logAudioConverter) << "Starting audioConverter() Input:" << inFormat.channelCount() << "Channels of" << inFormat.codec() << inFormat.sampleRate() << inFormat.sampleType() << inFormat.sampleSize() <<
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"Output:" << outFormat.channelCount() << "Channels of" << outFormat.codec() << outFormat.sampleRate() << outFormat.sampleType() << outFormat.sampleSize();
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if (inFormat.byteOrder() != outFormat.byteOrder()) {
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qInfo(logAudioConverter) << "Byteorder mismatch in:" << inFormat.byteOrder() << "out:" << outFormat.byteOrder();
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}
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if (inFormat.codec() == "audio/opus")
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{
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// Create instance of opus decoder
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int opus_err = 0;
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opusDecoder = opus_decoder_create(inFormat.sampleRate(), inFormat.channelCount(), &opus_err);
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qInfo(logAudioConverter()) << "Creating opus decoder: " << opus_strerror(opus_err);
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}
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if (outFormat.codec() == "audio/opus")
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{
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// Create instance of opus encoder
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int opus_err = 0;
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opusEncoder = opus_encoder_create(outFormat.sampleRate(), outFormat.channelCount(), OPUS_APPLICATION_AUDIO, &opus_err);
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//opus_encoder_ctl(opusEncoder, OPUS_SET_LSB_DEPTH(16));
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//opus_encoder_ctl(opusEncoder, OPUS_SET_INBAND_FEC(1));
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//opus_encoder_ctl(opusEncoder, OPUS_SET_DTX(1));
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//opus_encoder_ctl(opusEncoder, OPUS_SET_PACKET_LOSS_PERC(5));
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opus_encoder_ctl(opusEncoder, OPUS_SET_COMPLEXITY(opusComplexity)); // Reduce complexity to maybe lower CPU?
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qInfo(logAudioConverter()) << "Creating opus encoder: " << opus_strerror(opus_err);
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}
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if (inFormat.sampleRate() != outFormat.sampleRate())
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{
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int resampleError = 0;
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unsigned int ratioNum;
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unsigned int ratioDen;
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// Sample rate conversion required.
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resampler = wf_resampler_init(outFormat.channelCount(), inFormat.sampleRate(), outFormat.sampleRate(), resampleQuality, &resampleError);
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wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
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resampleRatio = static_cast<double>(ratioDen) / ratioNum;
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qInfo(logAudioConverter()) << "wf_resampler_init() returned: " << resampleError << " resampleRatio: " << resampleRatio;
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}
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return true;
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}
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audioConverter::~audioConverter()
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{
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qInfo(logAudioConverter) << "Closing audioConverter() Input:" << inFormat.channelCount() << "Channels of" << inFormat.codec() << inFormat.sampleRate() << inFormat.sampleType() << inFormat.sampleSize() <<
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"Output:" << outFormat.channelCount() << "Channels of" << outFormat.codec() << outFormat.sampleRate() << outFormat.sampleType() << outFormat.sampleSize();
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if (opusEncoder != Q_NULLPTR) {
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qInfo(logAudioConverter()) << "Destroying opus encoder";
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opus_encoder_destroy(opusEncoder);
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}
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if (opusDecoder != Q_NULLPTR) {
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qInfo(logAudioConverter()) << "Destroying opus decoder";
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opus_decoder_destroy(opusDecoder);
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}
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if (resampler != Q_NULLPTR) {
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speex_resampler_destroy(resampler);
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qDebug(logAudioConverter()) << "Resampler closed";
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}
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}
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bool audioConverter::convert(audioPacket audio)
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{
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// If inFormat and outFormat are identical, just emit the data back (removed as it doesn't then process amplitude)
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if (audio.data.size() > 0)
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{
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if (inFormat.codec() == "audio/opus")
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{
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unsigned char* in = (unsigned char*)audio.data.data();
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//Decode the frame.
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int nSamples = opus_packet_get_nb_samples(in, audio.data.size(), inFormat.sampleRate());
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if (nSamples == -1) {
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// No opus data yet?
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return false;
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}
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QByteArray outPacket(nSamples * sizeof(float) * inFormat.channelCount(), (char)0xff); // Preset the output buffer size.
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float* out = (float*)outPacket.data();
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//if (audio.seq > lastAudioSequence + 1) {
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// nSamples = opus_decode_float(opusDecoder, Q_NULLPTR, 0, out, nSamples, 1);
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//}
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//else {
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nSamples = opus_decode_float(opusDecoder, in, audio.data.size(), out, nSamples, 0);
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//}
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//lastAudioSequence = audio.seq;
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audio.data.clear();
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audio.data = outPacket; // Replace incoming data with converted.
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}
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else if (inFormat.codec() == "audio/PCMU")
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{
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// Current packet is "technically" 8bit so need to create a new buffer that is 16bit
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QByteArray outPacket((int)audio.data.length() * 2, (char)0xff);
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qint16* out = (qint16*)outPacket.data();
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for (int f = 0; f < audio.data.length(); f++)
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{
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*out++ = ulaw_decode[(quint8)audio.data[f]];
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}
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audio.data.clear();
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audio.data = outPacket; // Replace incoming data with converted.
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// Make sure that sample size/type is set correctly
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}
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Eigen::VectorXf samplesF;
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if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 32)
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{
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Eigen::Ref<VectorXint32> samplesI = Eigen::Map<VectorXint32>(reinterpret_cast<qint32*>(audio.data.data()), audio.data.size() / int(sizeof(qint32)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint32>::max());
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}
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else if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 16)
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{
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Eigen::Ref<VectorXint16> samplesI = Eigen::Map<VectorXint16>(reinterpret_cast<qint16*>(audio.data.data()), audio.data.size() / int(sizeof(qint16)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint16>::max());
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}
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else if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 8)
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{
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Eigen::Ref<VectorXint8> samplesI = Eigen::Map<VectorXint8>(reinterpret_cast<qint8*>(audio.data.data()), audio.data.size() / int(sizeof(qint8)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint8>::max());;
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}
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else if (inFormat.sampleType() == QAudioFormat::UnSignedInt && inFormat.sampleSize() == 8)
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{
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Eigen::Ref<VectorXuint8> samplesI = Eigen::Map<VectorXuint8>(reinterpret_cast<quint8*>(audio.data.data()), audio.data.size() / int(sizeof(quint8)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<quint8>::max());;
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}
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else if (inFormat.sampleType() == QAudioFormat::Float) {
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(audio.data.data()), audio.data.size() / int(sizeof(float)));
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}
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else
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{
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qInfo(logAudio()) << "Unsupported Input Sample Type:" << inFormat.sampleType() << "Size:" << inFormat.sampleSize();
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}
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if (samplesF.size() > 0)
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{
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audio.amplitude = samplesF.array().abs().maxCoeff();
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// Set the volume
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samplesF *= audio.volume;
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/*
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samplesF is now an Eigen Vector of the current samples in float format
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The next step is to convert to the correct number of channels in outFormat.channelCount()
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*/
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if (inFormat.channelCount() == 2 && outFormat.channelCount() == 1) {
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// If we need to drop one of the audio channels, do it now
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Eigen::VectorXf samplesTemp(samplesF.size() / 2);
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samplesTemp = Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesF.data(), samplesF.size() / 2);
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samplesF = samplesTemp;
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}
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else if (inFormat.channelCount() == 1 && outFormat.channelCount() == 2) {
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// Convert mono to stereo if required
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Eigen::VectorXf samplesTemp(samplesF.size() * 2);
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Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data(), samplesF.size()) = samplesF;
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Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data() + 1, samplesF.size()) = samplesF;
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samplesF = samplesTemp;
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}
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/*
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Next step is to resample (if needed)
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*/
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if (resampler != Q_NULLPTR && resampleRatio != 1.0)
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{
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quint32 outFrames = ((samplesF.size() / outFormat.channelCount()) * resampleRatio);
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quint32 inFrames = (samplesF.size() / outFormat.channelCount());
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QByteArray outPacket(outFrames * outFormat.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
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const float* in = (float*)samplesF.data();
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float* out = (float*)outPacket.data();
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int err = 0;
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if (outFormat.channelCount() == 1) {
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err = wf_resampler_process_float(resampler, 0, in, &inFrames, out, &outFrames);
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}
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else {
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err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
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}
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if (err) {
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qInfo(logAudioConverter()) << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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/*
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If output is Opus so encode it now, don't do any more conversion on the output of Opus.
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*/
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if (outFormat.codec() == "audio/opus")
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{
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float* in = (float*)samplesF.data();
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QByteArray outPacket(1275, (char)0xff); // Preset the output buffer size to MAXIMUM possible Opus frame size
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unsigned char* out = (unsigned char*)outPacket.data();
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int nbBytes = opus_encode_float(opusEncoder, in, (samplesF.size() / outFormat.channelCount()), out, outPacket.length());
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if (nbBytes < 0)
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{
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qInfo(logAudioConverter()) << "Opus encode failed:" << opus_strerror(nbBytes) << "Num Samples:" << samplesF.size();
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return false;
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}
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else {
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outPacket.resize(nbBytes);
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audio.data.clear();
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audio.data = outPacket; // Copy output packet back to input buffer.
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//samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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}
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else {
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/*
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Now convert back into the output format required
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*/
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audio.data.clear();
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if (outFormat.sampleType() == QAudioFormat::UnSignedInt && outFormat.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint8>::max());
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samplesITemp.array() += 127;
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VectorXuint8 samplesI = samplesITemp.cast<quint8>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(quint8)));
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}
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else if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint8>::max());
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VectorXint8 samplesI = samplesITemp.cast<qint8>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint8)));
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}
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else if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 16)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
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VectorXint16 samplesI = samplesITemp.cast<qint16>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
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}
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else if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 32)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint32>::max());
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VectorXint32 samplesI = samplesITemp.cast<qint32>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint32)));
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}
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else if (outFormat.sampleType() == QAudioFormat::Float)
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{
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audio.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
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}
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else {
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qInfo(logAudio()) << "Unsupported Output Sample Type:" << outFormat.sampleType() << "Size:" << outFormat.sampleSize();
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}
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/*
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As we currently don't have a float based uLaw encoder, this must be done
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after all other conversion has taken place.
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*/
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if (outFormat.codec() == "audio/PCMU")
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{
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QByteArray outPacket((int)audio.data.length() / 2, (char)0xff);
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qint16* in = (qint16*)audio.data.data();
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for (int f = 0; f < outPacket.length(); f++)
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{
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qint16 sample = *in++;
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int sign = (sample >> 8) & 0x80;
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if (sign)
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sample = (short)-sample;
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if (sample > cClip)
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sample = cClip;
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sample = (short)(sample + cBias);
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int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
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int mantissa = (sample >> (exponent + 3)) & 0x0F;
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int compressedByte = ~(sign | (exponent << 4) | mantissa);
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outPacket[f] = (quint8)compressedByte;
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}
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audio.data.clear();
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audio.data = outPacket; // Copy output packet back to input buffer.
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}
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}
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}
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else
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{
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qDebug(logAudioConverter) << "Detected empty packet";
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}
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}
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emit converted(audio);
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return true;
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}
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